Use this option to assign the manual-in-interrupt interruptible level to the agent.
Avaya Aura® Communication Manager Screen Reference
Release 8.1.x Issue 11
March 2021
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Contents
Chapter 1: Introduction.......................................................................................................... 16 Purpose................................................................................................................................16 Change history......................................................................................................................16
Chapter 2: AAR Digit Analysis Table and ARS Digit Analysis Table................................. 19
Chapter 3: AAR Digit Conversion Table and ARS Digit Conversion Table....................... 24
Chapter 4: Abbreviated Dialing............................................................................................. 27 Enhanced List....................................................................................................................... 27 Group List.............................................................................................................................29 Personal List.........................................................................................................................31 System List...........................................................................................................................32 7103A Button Assignment......................................................................................................34
Chapter 5: Access Endpoint.................................................................................................. 36
Chapter 6: Administered Connection................................................................................... 40 AUTHORIZED TIME OF DAY.................................................................................................41 MISCELLANEOUS PARAMETERS........................................................................................ 43
Chapter 7: Agent Login ID......................................................................................................45 Agent Login ID: page 1.......................................................................................................... 45 MWI Served User Type....................................................................................................50 Unicode Name................................................................................................................ 51 Agent Login ID: page 2.......................................................................................................... 52
Chapter 8: Alias Station......................................................................................................... 55
Chapter 9: Alphanumeric Dialing Table................................................................................ 56
Chapter 10: Announcements/Audio Sources.......................................................................57 Live Stream Source............................................................................................................... 59
Chapter 11: ARS Toll Table.................................................................................................... 63
Chapter 12: Attendant Console............................................................................................. 64 Attendant console: page 1......................................................................................................64 Attendant console: page 2 Softconsole IP Attendant................................................................ 69 Attendant console: page 2 VIS feature options.........................................................................71 Attendant console: page 3......................................................................................................71 Attendant console: page 3 feature button assignments............................................................. 74 Attendant console: page 4 display module button assignments................................................. 77
Chapter 13: Audio Group....................................................................................................... 78
Chapter 14: AUDIX-MSA Node Names.................................................................................. 80
Chapter 15: Authorization Code -- COR Mapping...............................................................81
Chapter 16: Best Service Routing......................................................................................... 83
Chapter 17: Bulletin board..................................................................................................... 86
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Chapter 18: Button Type Customization Restriction...........................................................87
Chapter 19: Call Type Digit Analysis Table.......................................................................... 89
Chapter 20: Call Vector...........................................................................................................91
Chapter 21: Calling Party Number Conversion for Tandem Calls......................................97 Len.................................................................................................................................... 100 Outgoing Number Format.....................................................................................................101 Outgoing Trunk Group(s)..................................................................................................... 101
Chapter 22: CAMA Numbering Format............................................................................... 102
Chapter 23: CDR system parameters..................................................................................104 CDR system parameters: page 1..........................................................................................104 CDR retention (days)..................................................................................................... 105 CDR System Parameters: page 2......................................................................................... 114
Chapter 24: Change Station Extension...............................................................................117
Chapter 25: Circuit Packs.................................................................................................... 120
Chapter 26: Class of Restriction......................................................................................... 121 Class of restriction: page 1...................................................................................................121 Class of Restriction: page 2..................................................................................................130 Class of Restriction: page 3..................................................................................................134 Allow Trunk Seizure On Busy Indicator............................................................................135 Class of restriction: page 4 to page 23.................................................................................. 138
Chapter 27: Class of service................................................................................................140 Class of service: page 1.......................................................................................................141 Class of service: page 2.......................................................................................................145 Bridging Exclusion Override........................................................................................... 145 DCC Activation/Deactivation...........................................................................................146
Chapter 28: Cluster Session Manager................................................................................ 148
Chapter 29: Code Calling IDs...............................................................................................150
Chapter 30: Configuration Set............................................................................................. 151 Call Log Notify.................................................................................................................... 153 EC500Delay Deactivation State............................................................................................155 Feature Invocation by In-Call DTMF Code.............................................................................156 Location for routing incoming overlap calls............................................................................ 156
Chapter 31: Console parameters.........................................................................................158 Console parameters: page 1................................................................................................ 158 SIP...............................................................................................................................163 Console parameters: page 2................................................................................................ 164 Console parameters: page 4................................................................................................ 170 Console parameters: page 5................................................................................................ 170
Chapter 32: Coverage Answer Group................................................................................. 171
Chapter 33: Coverage Path.................................................................................................. 173 Criteria............................................................................................................................... 175
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Coverage points..................................................................................................................177 Chapter 34: Crisis Alert System Parameters......................................................................179
ALERT PAGER................................................................................................................... 180 SNMP NOTIFICATION........................................................................................................ 183
SNMP Inform to Notify Adjunct When DCP and H.323 Stations Go In-Service....................183 SNMP Inform to Notify Adjunct When SIP Station Dials Emergency Call............................183 Chapter 35: CTI Link............................................................................................................. 184 CTI link: page 1...................................................................................................................184 CTI link: page 2...................................................................................................................186 IC Adjunct Routing...................................................................................................... 187 CTI link: page 3...................................................................................................................188 Bridged Appearance Origination Restriction.................................................................... 188 SAC/CF Override.......................................................................................................... 188 Chapter 36: Data Module......................................................................................................190 Data module: page 1........................................................................................................... 190 Data module: page 2: Type data-line.....................................................................................196 Data module: page 2 - Type 7500, WC-BRI, NI-BRI............................................................... 199 Chapter 37: Date and Time...................................................................................................201 Chapter 38: Daylight Saving Rules..................................................................................... 204 Chapter 39: DCS to QSIG TSC Gateway............................................................................. 208 Chapter 40: Dial Plan Analysis Table.................................................................................. 210 Chapter 41: Dial Plan Parameters....................................................................................... 214 Call Type Analysis Behavior When Extension Not Assigned....................................................215 Location-Based Call Type Analysis....................................................................................... 218 Chapter 42: Digit Absorption............................................................................................... 220 Chapter 43: Display Parameters..........................................................................................221 Chapter 44: DS1 Circuit Pack.............................................................................................. 224 DS1 Circuit Pack: page 1..................................................................................................... 224 DS1 circuit pack: page 2...................................................................................................... 240 Chapter 45: Duplicate Station..............................................................................................243 Chapter 46: Duplicate Vector............................................................................................... 244 Chapter 47: Enable File Transfer......................................................................................... 246 Chapter 48: Enable Session.................................................................................................247 Chapter 49: Extended Pick-Up Group.................................................................................248 Chapter 50: Extensions administered to have an MCT-Control button........................... 249 Chapter 51: Extensions to Call Which Activate Features by Name................................. 250 Chapter 52: Failover Group Domain Mapping....................................................................251 Chapter 53: Feature Access Code (FAC)............................................................................ 253 Feature Access Code: page 1.............................................................................................. 253 Feature Access Code: page 2.............................................................................................. 258
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Feature Access Code: page 3.............................................................................................. 264 Feature Access Code: page 4.............................................................................................. 268 Feature Access Code: page 5.............................................................................................. 272 Feature Access Code: page 6.............................................................................................. 276 Feature Access Code: page 7.............................................................................................. 277 Feature Access Code: page 8.............................................................................................. 278 Feature Access Code: page 9.............................................................................................. 280
WNDP PRECEDENCE ACCESS CODES....................................................................... 281 Destination Code Control............................................................................................... 281
Chapter 54: Feature-related system parameters............................................................... 283 Feature-related system parameters: page 1.......................................................................... 283 Feature-related system parameters: page 2.......................................................................... 290 Feature-related system parameters: page 3.......................................................................... 292 Feature-related system parameters: page 4.......................................................................... 295 Feature-related system parameters: page 5.......................................................................... 300 Feature-related system parameters: page 6.......................................................................... 304 Feature-related system parameters: page 7.......................................................................... 312 Feature-related system parameters: page 8.......................................................................... 316 ISDN PARAMETERS.....................................................................................................316 PARAMETERS FOR CREATING QSIG SELECTION NUMBERS......................................319 Feature-related system parameters: page 9.......................................................................... 321 Feature-related system parameters: page 10.........................................................................324 ITALIAN DCS PROTOCOL............................................................................................ 328 Feature-related system parameters: page 11......................................................................... 329 CALL CENTER SYSTEM PARAMETERS EAS................................................................329 VECTORING................................................................................................................ 330 Service Observing field descriptions................................................................................333 Feature-related system parameters: page 12.........................................................................335 AGENT AND CALL SELECTION.................................................................................... 335 ASAI ............................................................................................................................337 CALL MANAGEMENT SYSTEMS.................................................................................. 338 OTHER CALL MANAGEMENT SYSTEM FIELDS............................................................339 Feature-related system parameters: page 13.........................................................................341 Feature-related system parameters: page 14.........................................................................344 Feature-related system parameters: page 15.........................................................................347 FORCED AGENT LOGOUT PARAMETERS................................................................... 347 SIP STATION REACHABILITY CHECKING OPTIONS..................................................... 349 Feature-related system parameters: page 16.........................................................................350 Feature-related system parameters: page 17.........................................................................353 Feature-related system parameters: page 18.........................................................................356 Feature-related system parameters: page 19.........................................................................360 Commercial Cost Avoidance (CCA) Related System Parameters............................................ 365
Chapter 55: Firmware Station Download............................................................................367
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Chapter 56: Group Paging Using Speakerphone...............................................................370
Chapter 57: Holiday Table.................................................................................................... 374
Chapter 58: Hospitality.........................................................................................................377 Hospitality: page 1...............................................................................................................377 Hospitality: page 2...............................................................................................................381 Hospitality: page 3...............................................................................................................387
Chapter 59: Hunt Group....................................................................................................... 388 Hunt group: page 1..............................................................................................................389 SIP URI........................................................................................................................ 397 Hunt group: page 2..............................................................................................................398 Hunt group: page 3..............................................................................................................408 Interruptible Aux Deactivation Threshold......................................................................... 409 Hunt Group: page 4............................................................................................................. 411 Hunt Group: page 5 through X..............................................................................................417
Chapter 60: Incoming Call Handling Treatment................................................................. 420
Chapter 61: Integrated Announcement Boards................................................................. 424
Chapter 62: Integrated Announcement Translations........................................................ 426
Chapter 63: Intercom Group................................................................................................ 428
Chapter 64: Inter-Exchange Carrier (IXC) Codes............................................................... 430 Inter-Exchange Carrier (IXC) Codes: page 1..........................................................................430 Inter-Exchange Carrier (IXC) Codes: page 2..........................................................................430
Chapter 65: Intra-Switch CDR..............................................................................................432
Chapter 66: IP Address Mapping.........................................................................................433
Chapter 67: IP Codec Set..................................................................................................... 435 IP Codec Set: page 1...........................................................................................................435 Encrypted SRTCP......................................................................................................... 437 IP Codec Set: page 2...........................................................................................................439 Media Connection IP Address Type Preferences..............................................................442 Packet Size...................................................................................................................443 SIP 64K Data................................................................................................................ 443 IP Codec Set: page 3...........................................................................................................444
Chapter 68: IP Interfaces......................................................................................................449 IP Interface: page 1............................................................................................................. 449 IP Interface: page 2............................................................................................................. 454 IP Interfaces: page 3........................................................................................................... 457
Chapter 69: IP network region............................................................................................. 459 IP network region: page 1.................................................................................................... 459 NR Group..................................................................................................................... 459 IP network region: page 2.................................................................................................... 465 ALTERNATIVE NETWORK ADDRESS TYPES................................................................465 RTCP Reporting to Monitor Server Enabled.....................................................................465
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IP network region: page 3.................................................................................................... 467 IP network region: page 4.................................................................................................... 470
Chapter 70: IP Node Names................................................................................................. 475
Chapter 71: IP options system parameters........................................................................ 476 SYSTEM PARAMETERS IP-OPTIONS: page 1..................................................................... 476 SYSTEM PARAMETERS IP-OPTIONS: page 2..................................................................... 480 SYSTEM PARAMETERS IP-OPTIONS: page 3..................................................................... 482 ALTERNATIVE NETWORK ADDRESS TYPES................................................................482 SYSTEM PARAMETERS IP-OPTIONS: page 4..................................................................... 484 SYSTEM PARAMETERS IP-OPTIONS: page 5..................................................................... 486 BW Management Option................................................................................................486 Signaling-group of Primary SM BW Mgr.......................................................................... 486 Signaling-group of Secondary SM BW Mgr......................................................................486
Chapter 72: IP Routing......................................................................................................... 488
Chapter 73: IP Server Interface (IPSI) Administration....................................................... 491 IP Server Interface (IPSI) Administration: page 1................................................................... 491 IP Server Interface (IPSI) Administration: page 2................................................................... 496
Chapter 74: IP Services........................................................................................................ 498 IP Services: page 1............................................................................................................. 498 TLS Encryption............................................................................................................. 500 IP Services: page 3............................................................................................................. 500 IP Services: page 4............................................................................................................. 502
Chapter 75: ISDN Trunk Group............................................................................................ 504 ISDN Trunk Group: page 1...................................................................................................504 ISDN Trunk Group: page 2...................................................................................................516 ISDN Trunk Group: page 3...................................................................................................523 QSIG Trunk Group Options.................................................................................................. 535 ISDN Trunk Group: Administrable Timers.............................................................................. 539 Shared UUI Feature Priorities...............................................................................................540 CBC Service Trunk Group Allocation Plan Assignment Schedule............................................ 542 CBC Trunk Group Usage Allocation...................................................................................... 543 Wideband Support Options.................................................................................................. 544 ISDN Trunk Group: Group Member Assignments................................................................... 546
Chapter 76: ISDN-BRI Trunk Circuit Pack...........................................................................548 ISDN-BRI Trunk Circuit Pack: page 1, TN2185 circuit pack.....................................................548 ISDN-BRI Trunk Circuit Pack: page 1, TN556B or TN2198 circuit packs.................................. 552 ISDN-BRI Trunk Circuit Pack: page 2....................................................................................553
Chapter 77: Language translations.....................................................................................556
Chapter 78: Listed Directory Numbers............................................................................... 563
Chapter 79: Locations.......................................................................................................... 564
Chapter 80: Location Parameters........................................................................................568 Location Parameters: page 1................................................................................................568
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Contents
Convert ISDN Public Numbers to International for Display................................................ 569 Time Format on Terminals..............................................................................................572 Replace International Access Code with '+'......................................................................574 Location Parameters: page 2................................................................................................574 Location Parameters: page 3................................................................................................575 Chapter 81: Login Administration....................................................................................... 576 Chapter 82: Logging levels.................................................................................................. 577 Logging Levels: page 1........................................................................................................577 Logging Levels: page 2........................................................................................................578 Chapter 83: Loudspeaker Paging........................................................................................579 Chapter 84: Maintenance-related system parameters.......................................................583 Chapter 85: Media-Gateway................................................................................................. 584 Chapter 86: Media Server.....................................................................................................585 Signaling Group.................................................................................................................. 585 Voip Channel License Limit.................................................................................................. 585 Dedicated Voip Channel Licenses.........................................................................................586 Capacity Profile Max Sessions............................................................................................. 586 Capacity Profile MPU Estimate.............................................................................................586 Cluster Type....................................................................................................................... 586 Chapter 87: Message Waiting Indication Subscriber Number Prefixes...........................588 Chapter 88: Mode Code Related System Parameters....................................................... 590 Chapter 89: Modem Pool Group.......................................................................................... 593 Chapter 90: MOH Group....................................................................................................... 599 Chapter 91: Multifrequency-Signaling-Related Parameters............................................. 601 Multifrequency-Signaling-Related Parameters: page 1........................................................... 601 Multifrequency-Signaling-Related Parameters: page 2........................................................... 609 Multifrequency-Signaling-Related Parameters: page 3........................................................... 612 Multifrequency - Signaling- Related Parameters: page 4.........................................................617 Chapter 92: Multiple Level Precedence & Preemption (MLPP) Parameters....................621 ANNOUNCEMENTS............................................................................................................621 ASAC Enabled.............................................................................................................. 621 Attendant Queue Announcement....................................................................................622 PRECEDENCE CALLING-DIALED DIGIT ASSIGNMENT.......................................................623 Chapter 93: Music Sources..................................................................................................627 Chapter 94: Network Facilities.............................................................................................630 Chapter 95: Network Region Group.................................................................................... 631 Name................................................................................................................................. 631 Chapter 96: Node Number Routing..................................................................................... 632 Chapter 97: Numbering-Private Format..............................................................................633 Chapter 98: Numbering -- Public/Unknown Format......................................................... 636
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Contents
Chapter 99: Off-PBX Telephone Mobile Feature Extensions............................................ 639 Chapter 100: Partition Routing Table..................................................................................640 Chapter 101: Personal CO Line Group............................................................................... 641
Personal CO Line Group: page 1..........................................................................................641 Personal CO Line Group: page 2..........................................................................................651 Chapter 102: Pickup Group..................................................................................................652 Chapter 103: Policy Routing Table......................................................................................654 Chapter 104: Precedence Routing Digit Analysis Table................................................... 657 Destination Code Control..................................................................................................... 657 Percentage IPBO Blocked....................................................................................................658 Chapter 105: Precedence Routing Digit Conversion Table.............................................. 660 Chapter 106: PRI Endpoint...................................................................................................662 Chapter 107: Processor Channel Assignment...................................................................666 Chapter 108: QSIG to DCS TSC Gateway........................................................................... 670 Chapter 109: Reason Code Names..................................................................................... 672 Chapter 110: Remote Access...............................................................................................674 Chapter 111: Remote Call Coverage Table......................................................................... 678 Chapter 112: Remote Office................................................................................................. 680 Chapter 113: RHNPA Table...................................................................................................682 Chapter 114: Route Pattern..................................................................................................683 Used for SIP Stations.......................................................................................................... 692 Chapter 115: Security related system parameters.............................................................693 Security related system parameters: page 1.......................................................................... 693 Security related system parameters: page 2.......................................................................... 695 Chapter 116: Service Hours Table....................................................................................... 698 Chapter 117: Signaling group.............................................................................................. 700 Signaling group: page 1....................................................................................................... 700
Clustered...................................................................................................................... 702 Signaling group: page 2....................................................................................................... 722
Enable Failover Event Package Subscription...................................................................724 Failover/failback Signaling-group Pair............................................................................. 724 OPTIONS Request Parameters...................................................................................... 725 Signaling group: NCA-TSC Assignment page........................................................................ 726 Chapter 118: SIT Treatment for Call Classification............................................................728 Chapter 119: Site Data.......................................................................................................... 732 Chapter 120: Station............................................................................................................. 733 Station: Page 1................................................................................................................... 739 Attendant......................................................................................................................739 Button Module Pages.................................................................................................... 740 Buttons per Page.......................................................................................................... 740
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Contents
Set............................................................................................................................... 753 System ID.....................................................................................................................757 Terminal Number...........................................................................................................762 Unicode Name.............................................................................................................. 763 Station: Page 2................................................................................................................... 764 EC500 State................................................................................................................. 770 Station: Page 3................................................................................................................... 777 Bridging Tone for This Extension.................................................................................... 777 Do not Share Port with Unnamed Registration................................................................. 778 Enhanced Callr-Info display for 1-line display phones....................................................... 778 Offline Call Logging....................................................................................................... 780 Require Mutual Authentication if TLS.............................................................................. 780 Station: Page 4................................................................................................................... 781 Station :Page 5................................................................................................................... 784 Station: Page 6................................................................................................................... 785 Enable Reachability for Station Domain Control............................................................... 785 Primary Session Manager and secondary Session Manager.............................................786 Route Selection.............................................................................................................786 SIP URI........................................................................................................................ 786
Chapter 121: Stations With Off-PBX Telephone Integration............................................. 790 Stations With Off-PBX Telephone Integration: page 1............................................................. 790 Dual Mode.................................................................................................................... 791 Stations With Off-PBX Telephone Integration: page 2............................................................. 793 Stations With Off-PBX Telephone Integration: page 3............................................................. 795
Chapter 122: Survivable ARS Analysis Table.................................................................... 796
Chapter 123: Survivable Processor.................................................................................... 798 Survivable processor: page 1............................................................................................... 798 Survivable Processor: page 3 (IP Services page)...................................................................801 Priority with respect to Media Servers............................................................................. 802 Survivable Processor: page 4 (IP Services -- Session Layer Timers page).............................. 803
Chapter 124: System Capacity............................................................................................ 805
Chapter 125: System Configuration....................................................................................806
Chapter 126: System Parameters Call Coverage/Call Forwarding...................................807
Chapter 127: System Parameters Country Options.......................................................... 817
Chapter 128: System Parameters Customer Options....................................................... 823 System parameters customer options: page 1....................................................................... 823 System parameters customer options: page 2....................................................................... 825 System parameters customer options: page 3....................................................................... 827 Maximum Number of Avaya Media Server VoIP Channels................................................ 827 System parameters customer options: page 4....................................................................... 828 System parameters customer options: page 5....................................................................... 833 System parameters customer options: page 6....................................................................... 838
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Contents
System parameters customer options: page 7....................................................................... 842 System parameters customer options: page 8....................................................................... 846 System parameters customer options: page 9 (QSIG OPTIONAL FEATURES)........................ 848 System parameters customer options: page 10 (ASAI FEATURES).........................................849 System parameters customer options: page 11 (MAXIMUM IP REGISTRATIONS BY PRODUCT ID).................................................................................................................... 850 Chapter 129: System parameters - Duplication................................................................. 852 Chapter 130: System parameters maintenance................................................................. 853 Chapter 131: System Parameters Media Gateway Automatic Recovery Rule................ 854 Chapter 132: Media Server Recovery Rules.......................................................................856 Auto Return........................................................................................................................ 856 Day.................................................................................................................................... 857 Report Expiration (sec)........................................................................................................ 857 Report Interval (sec)............................................................................................................ 857 Time...................................................................................................................................857 Time Delay......................................................................................................................... 858 Chapter 133: System parameters OCM call classification................................................859 System parameters OCM call classification: page 1............................................................... 859 System parameters OCM call classification: page 2............................................................... 860 Chapter 134: System Parameters Port Networks.............................................................. 862 System parameters port networks: page 1.............................................................................862 System parameters port networks: page 2.............................................................................863 Chapter 135: System Parameters - SCCAN........................................................................865 Chapter 136: System Parameters Special Applications....................................................867 Chapter 137: Telecommuting Access................................................................................. 868 Chapter 138: Tenant..............................................................................................................869 Tenant: page 1.................................................................................................................... 869
SIP...............................................................................................................................871 Tenant: page 2.................................................................................................................... 873 Chapter 139: Terminal Parameters......................................................................................875 Chapter 140: Terminating Extension Group.......................................................................880 Chapter 141: TFTP Server.................................................................................................... 884 Chapter 142: Time of Day Coverage Table......................................................................... 886 Chapter 143: Time of Day Routing Plan..............................................................................888 Chapter 144: Time of Day Station Lock Table.................................................................... 890 Chapter 145: Toll Analysis................................................................................................... 892 Chapter 146: Tone Generation.............................................................................................896 Tone Generation: page 1......................................................................................................896 Tone generation: page 2 (TONE GENERATION CUSTOMIZED TONES page).........................897 Chapter 147: Trunk Group....................................................................................................901
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Contents
Trunk Group........................................................................................................................901 Trunk Group: page 1......................................................................................................901 Trunk Group: page 2......................................................................................................920 Trunk Group: page 3......................................................................................................938
Administrable Timers........................................................................................................... 954 Trunk Group: ATMS Thresholds............................................................................................967 Trunk Group: Group Member Assignments............................................................................973 Trunk Group: Protocol Variations.......................................................................................... 976
Resend Display UPDATE Once on Receipt of 481 Response........................................... 977 Build Refer-To URI of REFER From Contact For NCR......................................................978 Request URI Contents................................................................................................... 981
Chapter 148: Uniform Dial Plan Table................................................................................. 983
Chapter 149: User Profile..................................................................................................... 986
Chapter 150: Variables for Vectors......................................................................................987
Chapter 151: Vector Directory Number...............................................................................990 Vector Directory Numbers: page 1........................................................................................ 990 SIP URI........................................................................................................................ 994 Unicode Name.............................................................................................................. 994 Vector Directory Number: page 2 (Meet-me Conference disabled)...........................................996 Used for BSR Polling................................................................................................... 1001 Vector Directory Number: page 2 (Meet-me Conference enabled)......................................... 1003 Vector Directory Number: page 3........................................................................................1005
Chapter 152: Video Bridge................................................................................................. 1008
Chapter 153: Virtual MAC Addresses............................................................................... 1012
Chapter 154: Resources.....................................................................................................1013 Communication Manager documentation.............................................................................1013 Finding documents on the Avaya Support website......................................................... 1015 Accessing the port matrix document............................................................................. 1016 Avaya Documentation Center navigation....................................................................... 1016 Training............................................................................................................................ 1017 Viewing Avaya Mentor videos.............................................................................................1018 Support............................................................................................................................ 1018 Using the Avaya InSite Knowledge Base....................................................................... 1018
Appendix A: PCN and PSN notifications.......................................................................... 1020 PCN and PSN notifications.................................................................................................1020 Viewing PCNs and PSNs................................................................................................... 1020 Signing up for PCNs and PSNs.......................................................................................... 1021
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Chapter 1: Introduction
Purpose
This document provides detailed description of the fields of Avaya Aura® Communication Manager Communication Manager screens. The primary audience for this document is:
· Avaya field technicians · Avaya partners · Technical support personnel · Solution architects · Implementation engineers · Support personnel · Technical support representatives
Change history
Issue 11 10
9
Date March 2021
February 2021
November 2020
Summary of changes Added the following section: · IC Adjunct Routing In Release 8.1.3.1, updated the following sections: · Site Data · Direct IP-IP Audio Connection Updated the "Support Request History" section.
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Change history
Issue 8 7
6 5
Date November 2020 October 2020
June 2020 April 2020
Summary of changes · Updated the "011000" section under the "Remote Call Coverage Table"
chapter. · Updated the "Station" section. · Added EC500 State field in "Station" chapter. In Release 8.1.3, added the following sections: · Unicode Name field for agent login · Resend Display UPDATE Once on Receipt of 481 Response · SNMP Inform to Notify Adjunct When DCP and H.323 Stations Go In-
Service · SNMP Inform to Notify Adjunct When SIP Station Dials Emergency Call · Do not Share Port with Unnamed Registration Updated the following sections: · Hunt group · Hold/Unhold notification · Trunk group protocol variation form image Updated the following images: · Feature-related system parameters: page 11 of 19 · Crisis alert system parameters · Station: page 3 of 5 Updated the "Call Limit" section. Added the following sections: · Capacity Profile Max Sessions · Capacity Profile MPU Estimate · Cluster Type
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Introduction
Issue 4
3 2 1
Date March 2020
October 2019 August 2019 June 2019
Summary of changes In Release 8.1.2, added the following sections: · CDR Retention (Days) · TLS Encryption · Busy/Reorder/Intercept Tone Timeout (Seconds) · MWI Served User Type In Release 8.1.2, updated the following sections: · CDR system parameters: page 1 · IP Services: page 1 · CMS (appl mis) · Meet-me Conference Access Code Change · Conference Type In Release 8.1.2, removed the following sections from the "Media Server" chapter: · Active-Spkr/MSS Video Channel License Limit · Dedicated Active-Spkr/MSS Video Channel License Limit · Composite Video Channel License Limit · Dedicated Composite Video Channel License Limit In Release 8.1.2, removed the following sections from the "System parameters customer options: Page 3" of "System Parameters Customer Options" chapter: · Maximum Number of AMS Active-Spkr/MSS Video Channels · Maximum Number of AMS Composite Video Channels Replaced the "Service observing" section with "Service Observing field descriptions" section. Updated the "Number of Members" section. Updated the "Tone Name" section. Release 8.1.
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Chapter 2: AAR Digit Analysis Table and ARS Digit Analysis Table
Communication Manager compares dialed numbers with the value of the Dialed Strings field and determines the route pattern for the dialed number.
To change the value of the fields on the AAR Digit Analysis Table screen, use the following command:
change aar analysis n, where n is the digit string.
To change the value of the fields on the ARS Digit Analysis Table screen, use the following command:
change ars analysis n, where n is the digit string.
ANI Reqd
Use this field to specify whether the R2-MFC and the Russian MF ANI calls require Automatic Number Identification (ANI).
The system displays the ANI Reqd field only if you disable the Request Incoming ANI (nonAAR/ARS) field on the Multifrequency-Signaling-Related Parameters screen.
Valid entry n y r
Usage
The R2-MFC or Russian MF ANI calls do not require ANI. The default value is n.
The R2-MFC or Russian MF ANI calls require ANI.
The call on the Russian Shuttle trunk or the Russian Rotary trunk drops if the ANI request fails.
The system displays this option only if you enable the Allow ANI Restriction on AAR/ARS field on the Feature-related system parameters screen.
Related links Allow ANI Restriction on AAR/ARS on page 305 Request Incoming ANI (non-AAR/ARS) on page 607
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AAR Digit Analysis Table and ARS Digit Analysis Table
Call Type (AAR only)
Use this field to enter the call type associated with each dialed string.
Valid entry aar intl
pubu
lev0lev2 unku
Usage
Use this option for regular Automatic Alternate Routing (AAR) calls.
Use this option if the Route index contains public network ISDN trunks that require international type of number encoding.
Use this option if the Route index contains public network ISDN trunks that require unknown type of number encoding.
Use this option for ISDN Private Numbering Plan (PNP) number formats.
Use this option to set up an Implicit numbering plan. Users can use the plan to dial an extension without using an Automatic Route Selection (ARS) or AAR access code.
ISDN Protocol:
Call type numbering aar intl pubu lev0 lev1 lev2
Numbering plan identifier E.164 (1) E.164 (1) E.164 (1) PNP(9) PNP(9) PNP(9)
Type of numbering national(2) international(1) unknown(0) local(4) Regional Level 1(2) Regional Level 2(1)
Call Type (ARS only)
Use this field to enter the call type associated with each dialed string.
Valid entry
Usage
intl alrt
emer fnpa
hpna lop
Use this option for public-network international calls.
Use this option to alert attendant consoles or other digital telephones when a user places an emergency call.
Use this option for emergency calls.
Use this option for ten-digit North American Numbering Plan (NANP) calls.
Use this option for seven-digit NANP calls.
Use this option for international operator calls.
China Number 1, Call Type toll-auto normal
normal attendant
normal attendant
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Valid entry
locl lpvt natl npvt nsvc op pubu
svcl svct svfl svft
Dialed String
Usage
Use this option for public-network local calls. Use this option local private calls. Use this option for non-NANP calls. Use this option for national private calls. Use this option for national service calls. Use this option for operator calls. Use this option for public-network number (E.164)-unknown calls. Use this option for national(2) calls. Use this option for national(2) calls. Use this option for service call first party control calls. Use this option for service call first party control calls.
China Number 1, Call Type normal normal normal normal normal attendant normal
toll-auto normal toll local
Dialed String
Communication Manager matches the dialed numbers with the entry in the Dialed String field that most closely matches the dialed number. You can enter up to 18 digits that the call-processing server analyzes. You can also enter the wildcard characters, x and X.
Location
This field displays the location of the telephone that is used to dial the number. For more
information about assigning a location and for a list of features that use the location, see Avaya Aura® Communication Manager Feature Description and Implementation.
Valid entry
1 to 250 or 1 to 2000
all
Usage
If there are matching entries in the location of the telephone, the system displays the matched entry.
If there are no matching entries in the location of the telephone, the system displays all.
Related links Multiple Locations on page 838
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AAR Digit Analysis Table and ARS Digit Analysis Table
Max
Use this field to specify the maximum number of user-dialed digits that the system must collect to match the dialed string.
Min
Use this field to specify the minimum number of user-dialed digits that the system must collect to match the dialed string.
Node Number
Use this field to specify a destination node number when the system uses node number routing or Distributed Communication System (DCS).
Valid entry blank 1 to 999
Usage Use this option if you do not want to enter the destination node number. Use this option to enter the destination node number.
Percent Full
This field displays the percentage of system memory resources used for the AAR and the ARS digit analysis. For more information about the system memory resources, see Avaya Aura® Communication Manager System Capacities Table.
Route Pattern
Use this field to specify the route number that Communication Manager must use for the dialed string.
Valid entry p1 to p2000 digits
Usage
To enter a route index number. To enter a route pattern. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
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Valid entry r1 to r32
node deny
Route Pattern
Usage To enter a Remote Home Numbering Plan Area (RHNPA) number if the dialed string requires RHNPA translations. To enable the system to use node number routing. To enable the system to block a call.
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Chapter 3: AAR Digit Conversion Table and ARS Digit Conversion Table
Use the AAR Digit Conversion Table and the ARS Digit Conversion Table screens to modify a dialed number for efficient routing. You can insert or delete digits from the dialed number. For example, you can delete 1 and the area code from the dialed number and avoid long distance charges by routing the call over the private network.
To change the value of the fields on the AAR Digit Conversion Table screen, use the following command:
change aar digit-conversion n, where n is the digit string.
To change the value of the fields on the ARS Digit Conversion Table screen, use the following command:
change ars digit-conversion n, where n is the digit string.
ANI Reqd
Use this field to specify whether the R2-MFC and the Russian MF ANI calls require Automatic Number Identification (ANI).
The system displays the ANI Reqd field only if you disable the Request Incoming ANI (nonAAR/ARS) field on the Multifrequency-Signaling-Related Parameters screen.
Valid entry n y r
Usage
The R2-MFC or Russian MF ANI calls do not require ANI. The default value is n.
The R2-MFC or Russian MF ANI calls require ANI.
The call on the Russian Shuttle trunk or the Russian Rotary trunk drops if the ANI request fails.
The system displays this option only if you enable the Allow ANI Restriction on AAR/ARS field on the Feature-related system parameters screen.
Related links Allow ANI Restriction on AAR/ARS on page 305 Request Incoming ANI (non-AAR/ARS) on page 607
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Conv
Use this field to enable additional digit conversion.
Conv
Del
Use this field to enter the number of digits that the system must delete from the starting of the dialed string.
Location
This field displays the location of the telephone that is used to dial the number. For more
information about assigning a location and for a list of features that use the location, see Avaya Aura® Communication Manager Feature Description and Implementation.
Valid entry
1 to 250 or 1 to 2000
all
Usage
If there are matching entries in the location of the telephone, the system displays the matched entry.
If there are no matching entries in the location of the telephone, the system displays all.
Related links Multiple Locations on page 838
Matching Pattern
Use this field to specify the pattern that the system must use to match the dialed number. The pattern can be up to 18 characters long and consist of digits and the wildcard characters, x and X.
Max
Use this field to specify the maximum number of user-dialed digits that the system must collect to match the dialed string.
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AAR Digit Conversion Table and ARS Digit Conversion Table
Min
Use this field to specify the minimum number of user-dialed digits that the system must collect to match the dialed string.
Net
Use this field to specify the method that the call processing server must use to analyze the converted number.
Valid entry ext aar ars
Usage The call processing server analyzes the converted number as an extension number. The call processing server analyzes the converted number as an AAR number. The call processing server analyzes the converted number as an ARS number.
Percent Full
This field displays the percentage of system memory resources used for the AAR and the ARS digit analysis. For more information about the system memory resources, see Avaya Aura® Communication Manager System Capacities Table.
Replacement String
Use this field to specify the digit string that must replace the deleted portion of the dialed number.
Valid entry blank 0 to 9, *
#
Usage To delete the digits without replacement. To type the digit string. The digit string can have up to 18 characters. To indicate the end of dialing.
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Chapter 4: Abbreviated Dialing
Use the Abbreviated Dialing screen to store and retrieve dialing lists that simplify or eliminate dialing. The types of Abbreviated Dialing lists are:
· Enhanced list · Group list · Personal list · System list Users can use these lists to: · Make local, long distance, and international calls. · Activate or deactivate features. · Access the remote system. Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31 System List on page 32 7103A Button Assignment on page 34
Enhanced List
Use this screen to create systemwide or personal lists for speed dialing.
Note: You must activate dialing in the license file before the system programs Abbreviated Dialing Enhanced List. To modify the value of the fields on the Enhanced List screen, use the following command: change abbreviated-dialing enhanced n, where n is an enhanced list number. Related links Abbreviated Dialing on page 27
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Abbreviated Dialing
DIAL CODE on page 28 Privileged on page 28 Size (multiple of 5) on page 28 Abbreviated Dialing Enhanced List on page 828
DIAL CODE
Use this field to specify the number that the system must dial when a user enters the dial code. The number can have up to 24 characters. You can also assign a feature access code.
Valid entry 0 to 9 * or # ~p ~w ~m ~s ~W
Usage To specify the number. To specify the feature access code. To enable the system to pause for 1.5 seconds. To enable the system to wait for dial tone. To enable the system to outpulse DTMF digits at the end-to-end rate. To enable the system to suppress the display of the digits being outpulsed. To enable the system to wait indefinitely for the dial tone if the network response time is more than 30 seconds. This entry is unavailable for S8300E servers.
Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31 System List on page 32
Privileged
Use this field to allow users to dial any number in the list, regardless of the class of restriction number. The default value is n.
Related links Enhanced List on page 27 Group List on page 29 System List on page 32
Size (multiple of 5)
Use this field to specify the number of dial code entries required in the list. You can enter a value from 5 to 100 in multiples of 5. The default value is 5.
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Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31 System List on page 32
Group List
Group List
Use this screen to create a group list. You can assign up to 100 numbers to every group list.
To change the value of the fields on the Group List screen, use the following command:
change abbreviated-dialing group n, where n is the group list number.
Related links Abbreviated Dialing on page 27 DIAL CODE on page 28 Group List on page 30 Privileged on page 28 Program Ext on page 30 Size (multiple of 5) on page 28
DIAL CODE
Use this field to specify the number that the system must dial when a user enters the dial code. The number can have up to 24 characters. You can also assign a feature access code.
Valid entry 0 to 9 * or # ~p ~w ~m ~s ~W
Usage To specify the number. To specify the feature access code. To enable the system to pause for 1.5 seconds. To enable the system to wait for dial tone. To enable the system to outpulse DTMF digits at the end-to-end rate. To enable the system to suppress the display of the digits being outpulsed. To enable the system to wait indefinitely for the dial tone if the network response time is more than 30 seconds. This entry is unavailable for S8300E servers.
Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31
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Abbreviated Dialing
System List on page 32
Group List
This field displays the number that you assign to the group list. Related links
Group List on page 29
Privileged
Use this field to allow users to dial any number in the list, regardless of the class of restriction number. The default value is n. Related links Enhanced List on page 27 Group List on page 29 System List on page 32
Program Ext
Use this field to enter the extension that has permissions to program the Group list. Related links
Group List on page 29
Size (multiple of 5)
Use this field to specify the number of dial code entries required in the list. You can enter a value from 5 to 100 in multiples of 5. The default value is 5. Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31 System List on page 32
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Personal List
Personal List
Use this screen to create a personal dialing list for telephone or data module users. To change the value of the fields on the Personal List screen, use the following command: change abbreviated-dialing personal n, where n is a personal list number. Related links Abbreviated Dialing on page 27 DIAL CODE on page 28 List Number on page 31 Personal List on page 32 Size (multiple of 5) on page 28
DIAL CODE
Use this field to specify the number that the system must dial when a user enters the dial code. The number can have up to 24 characters. You can also assign a feature access code.
Valid entry 0 to 9 * or # ~p ~w ~m ~s ~W
Usage To specify the number. To specify the feature access code. To enable the system to pause for 1.5 seconds. To enable the system to wait for dial tone. To enable the system to outpulse DTMF digits at the end-to-end rate. To enable the system to suppress the display of the digits being outpulsed. To enable the system to wait indefinitely for the dial tone if the network response time is more than 30 seconds. This entry is unavailable for S8300E servers.
Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31 System List on page 32
List Number
This field displays which of the three personal lists is defined for the telephone. Related links
Personal List on page 31
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Abbreviated Dialing
Personal List
Use this field to enter the extension of the telephone using this list. Related links
Personal List on page 31
Size (multiple of 5)
Use this field to specify the number of dial code entries required in the list. You can enter a value from 5 to 100 in multiples of 5. The default value is 5. Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31 System List on page 32
System List
Use this screen to create the system list. You can create only one system list. To change the value of the fields on the System List screen, use the following command: change abbreviated-dialing system n, where n is the system list number. Related links Abbreviated Dialing on page 27 DIAL CODE on page 28 Label Language on page 33 LABELS FOR 2420/4620 STATIONS on page 34 Privileged on page 28 Size (multiple of 5) on page 28
DIAL CODE
Use this field to specify the number that the system must dial when a user enters the dial code. The number can have up to 24 characters. You can also assign a feature access code.
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System List
Valid entry 0 to 9 * or # ~p ~w ~m ~s ~W
Usage To specify the number. To specify the feature access code. To enable the system to pause for 1.5 seconds. To enable the system to wait for dial tone. To enable the system to outpulse DTMF digits at the end-to-end rate. To enable the system to suppress the display of the digits being outpulsed. To enable the system to wait indefinitely for the dial tone if the network response time is more than 30 seconds. This entry is unavailable for S8300E servers.
Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31 System List on page 32
Label Language
Use this field to administer custom labels on the 2420 and 4620 telephones. The system can display the labels in one of the following languages:
· english
· french
· italian
· spanish
Note:
Unicode is not a valid language.
If you change the language, the system saves all administered labels in the new language and displays the labels in the new language.
To translate the labels to a different language, type user-defined in the Label Language field. You can use Avaya Message Editing Tool (AMET) to assign translations to the labels. For more information about AMET, see Job Aid: Telephone Message Translations.
Related links System List on page 32 Display Character Set on page 821
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Abbreviated Dialing
LABELS FOR 2420/4620 STATIONS
Use this field to customize the labels for the Abbreviated Dialing buttons on the 2420 and 4620 telephones. The labels can have up to 15 alphanumeric characters. Related links System List on page 32
Privileged
Use this field to allow users to dial any number in the list, regardless of the class of restriction number. The default value is n. Related links Enhanced List on page 27 Group List on page 29 System List on page 32
Size (multiple of 5)
Use this field to specify the number of dial code entries required in the list. You can enter a value from 5 to 100 in multiples of 5. The default value is 5. Related links Enhanced List on page 27 Group List on page 29 Personal List on page 31 System List on page 32
7103A Button Assignment
Use this screen to assign abbreviated dialing numbers to the 7103A telephone buttons. You can use this list only for 7103A fixed feature telephones. You can implement only one 7103A abbreviated dialing list and this list applies to all 7103A fixed feature telephones in the system. To modify the value of the fields on the 7103A Button Assignment screen, use the following command: change abbreviated-dialing 7103A-buttons n where n is a 7103A button assignment abbreviated dialing list.
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Related links Abbreviated Dialing on page 27 DIAL CODE on page 35
7103A Button Assignment
DIAL CODE
Use this field to assign a number to each dial code button. Any additions or changes apply to all 7103A fixed feature telephones. A call progress tone receiver is tied up while the system is waiting. The outgoing calling capability is impaired as there are a limited number of receivers in the system.
You can also assign a VDN.
Valid entry 0 to 9 * or # ~p ~w ~m ~s ~W
Usage To specify the number. To specify the feature access code. To enable the system to pause for 1.5 seconds. To enable the system to wait for dial tone. To enable the system to outpulse DTMF digits at the end-to-end rate. To enable the system to suppress the display of the digits being outpulsed. To enable the system to wait indefinitely for the dial tone if the network response time is more than 30 seconds. This entry is unavailable for S8300E servers.
Related links 7103A Button Assignment on page 34
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Chapter 5: Access Endpoint
Use the Access Endpoint screen to administer an access endpoint or a Wideband Access Endpoint (WAE).
WAE is an endpoint application connected to line-side non-ISDN T1 or E1 facilities. WAEs do not have signaling interface with the system. You can administer WAE only if Wideband Switching is enabled for the system.
An access endpoint is a nonsignaling trunk that neither responds to signaling nor generates signaling. You can use access endpoints to assign an extension number to a single trunk and eliminate the need to dedicate an entire trunk group for a single trunk. You can specify an access endpoint as the originator or the destination endpoint of an administered connection.
Note:
Access endpoints and WAEs use the same resources that the trunks use. Therefore, the sum of access endpoints and trunks cannot exceed the maximum number of trunks available in the system.
To change the value of the fields on the Access Endpoint screen, use the following command:
add access-endpoint next, where next is the next available extension.
Related links Wideband Switching on page 842
Communication Type
Use this field to specify the communication type for the access endpoint.
Valid entry 56k-data
64K-data voice-grade-data wideband
Usage For a DS1 access endpoint. You can use this option for the robbed-bit trunks. The default value is 56k-data. For a DS1 access endpoint. Do not use this option for the robbed-bit trunks. For an analog tie trunk access endpoint. For a WAE.
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COR
COR
Use this field to assign a class of restriction (COR) number to the access endpoint. You must administer this field so that only an Administered Connection (AC) endpoint can be connected to another AC endpoint.
COS
Use this field to assign a class of service (COS) number to the access endpoint. You must administer this field so that Communication Manager restricts the use of the Call Forwarding All Calls feature on the access endpoint.
Extension
Use this field to assign an extension number to the nonsignaling trunk.
ITC
Use this field to assign a transmission facility to the access endpoint. The system displays this field only if the Communication Type field is set to 56k-data, 64k-data, or Wideband.
Valid entry restricted
unrestricted
Usage
Restricted transmission facilities are used to complete the call. Before using this option, ensure that the associated port of the access endpoint is a channel of a DS1 circuit pack and the Line Coding field of the DS1 circuit pack is set to B8ZS. The default value is restricted.
Unrestricted transmission facilities are used to complete the call.
Name
Use this field to provide a name for the access endpoint.
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
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Access Endpoint
(Starting) Port
Use this field to assign a port number to the access endpoint. You can assign the port number by using one the following formats:
· cabinet number : carrier number : slot number : circuit number
Valid entry
01 to 64
A to E
01 to 20
01 to 04 (Analog TIE trunks)
01 to 31
Usage The first and second characters represent the cabinet number. The third character represents the carrier number. The fourth and fifth characters represent the slot number. The sixth and seventh characters represent the circuit number.
· gateway number : module number : circuit number
Valid entry 001 to 250 V1 to V9 01 to 31
Usage The first three characters represent the gateway number. The fourth and fifth characters represent the module number. The sixth and seventh characters represent the circuit number.
Note:
For WAEs, use the DS1 circuit packs, not analog tie trunks.
Related links Signaling Mode on page 237
TN
Use this field to assign a tenant number to the access endpoint. You can enter a value from 1 to 250.
Width
The system displays this field only if the Communication Type field is set to wideband. Use this field to provide the number of adjacent DS0 ports beginning with the specified starting port that make the WAE. You can enter a value between 2 to 31. The default entry is 6.
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Related links Communication Type on page 36
Width
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Chapter 6: Administered Connection
Use the Administered Connection screen to assign an end-to-end Administered Connection (AC) between two access endpoints or data endpoints. The system establishes the AC after a restart or when the AC is active. To change the value of the fields on the Administered Connection screen, use the following command: change administered-connection n, where n is the AC number.
Connection Number
This field displays the AC number.
Destination
Use this field to specify the address of the destination access endpoint or data endpoint. The endpoint can be remotely connected to the server on which the AC is assigned. You can enter one of the following:
· A local extension that is assigned to an access endpoint or a data endpoint · An Abbreviated Dialing entry The value must be consistent with the dial plan.
Enable
Use this field to enable Communication Manager to establish the AC when the AC is active.
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Name
Valid entry Usage
y
Communication Manager attempts to establish the AC when the AC is active. The default
value is y.
n
Communication Manager does not attempt to establish the AC when the AC is active.
Name
Use this field to enter the name of the AC. You can enter up to 27 alphanumeric characters.
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Originator
Use this field to specify an assigned access endpoint extension or data module extension. The endpoint must be registered with the same instance of Communication Manager on which you administered the nonsignaling DS1 trunk or the analog tie trunk.
AUTHORIZED TIME OF DAY
Continuous
Use this field to schedule the AC to be active for the entire time. The AC is reestablished if the connection fails.
Valid entry
y
n
Usage
The AC is active for the entire time. The default value is y. The AC is active only for the scheduled days and times.
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Administered Connection
Duration
Use this field to specify the duration for which the scheduled AC must remain active. The maximum duration is 167:59. The system displays this field only if you set the Continuous field to n.
Valid entry 000 to 167 00 to 59
Usage To enter the duration in hours. To enter the duration in minutes.
Start Days (Sun through Sat)
Use the following fields to specify the days when an attempt is made to establish the AC: · Sun · Mon · Tue · Wed · Thu · Fri · Sat
The system displays these fields only if you set the Continuous field to n. The start days can be the days when the AC is inactive. A scheduled AC might be active over a number of days. In this situation, use the fields only to specify the days when the AC starts. Related links Continuous on page 41
Start Time
Use this field to specify the time of day when Communication Manager must establish the AC. You can enter a value from 00:00 to 23:59. The system displays this field only if you set the Continuous field to n. Related links Continuous on page 41
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MISCELLANEOUS PARAMETERS
MISCELLANEOUS PARAMETERS
Alarm Threshold
Use this field to specify the number of attempts to establish or reestablish an AC that must fail consecutively before the system generates an AC alarm. For example, If you set this field to 1, the system generates an alarm after the first failure. You can enter a value from 1 to 10. The default value is 5.
The system displays this field only if you set the Alarm Type field to a value other than none.
Related links Alarm Type on page 43
Alarm Type
Use this field to specify the type of alarm generated when: · The AC cannot be established. · The AC cannot be reestablished after a failure. · The number of consecutive failures equals the alarm threshold.
Valid entry warning major minor
none
Usage For failures that do not cause significant degradation of service. These alarms are not reported to the attendant console or Initialization and Administration System (INADS). For failures that cause critical degradation of service and require immediate attention. For failures that:
· Cause degradation of service, but continue to support a crucial part of the system.
· Require action, but the consequences of the failure are gradual.
· Interfere with a feature or impair the services to trunks or stations. For disabling the alarm notification.
Auto Restoration
Use this field to enable the system to reestablish the AC that failed. The Auto restoration field is available only for the AC that is established over an ISDN Software Defined Data Network (SDDN) trunk group.
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Administered Connection
Priority
Use this field to specify the order in which the ACs are established. You must enter a value from 1 to 8. The highest priority is 1, and the lowest priority is 8. The default priority is 5.
Retry Interval
Use this field to specify the duration between the attempts to establish or reestablish the AC. You must enter the duration in minutes. The default value is 2 minutes.
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Chapter 7: Agent Login ID
In an Expert Agent Selection (EAS) environment, use the Agent Login ID screen to: · Add an agent login ID. · Change the options of an agent login ID. · Assign skills to an agent login ID.
To change the value of fields on the Agent Login ID screen, use one of the following commands: · change agent-loginID n, where n is the agent login ID. For any changes to take effect, the agent must log out and log in again. · change agent-loginID n auto, where n is the agent login ID. For the changes related to the skill set to take effect, the system logs out and logs in the agent. However, for the other changes to take effect, the agent must log out and log in again.
Agent Login ID: page 1
AAS
Use this field to enable the system to use the agent login ID as a port for Auto Available Split/Skill (AAS).
Valid entry n y
Usage
To use the agent login ID as a port for the switch adjunct equipment.
To use the agent login ID as a port for AAS. If you use this option, you must run the remove Agent-LoginID command.
ACW Agent Considered Idle
Use this field to include the After Call Work (ACW) agents in the Most-Idle Agent (MIA) queue.
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Agent Login ID
Valid entry system n y
Usage To apply the systemwide values. To exclude the ACW agents from the MIA queue. To include the ACW agents in the MIA queue.
Related links ACW Agents Considered Idle on page 335
AUDIX
Use this field to enable the system to use the agent login ID as a port for voice messaging.
If an agent login ID is used as a port for AAS, the agent login ID cannot be used as a port for voice messaging.
Audix Name for Messaging
Use this field to assign a name to the messaging system that is used for Leave Word Calling (LWC) reception. The messaging system also provides call coverage to the agent login ID.
Auto Answer
Use this field to enable the auto answer setting of the agent login ID.
Valid entry station acd
none all
Usage
The auto answer setting on the station overrides the auto answer setting of the agent.
The auto answer setting applies only to the ACD and direct agent calls. The agent extension gets a ringing tone for non-ACD calls.
The auto answer setting is disabled. The agent extension gets a ringing tone for all calls.
The auto answer setting applies to all ACD and non-ACD calls to the agent. For nonACD calls, the agent extension receives a single ring. To prevent the ring, use the ringeroff button.
Related links Allow Ringer-off with Auto-Answer on page 341 Auto Answer on page 765
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Agent Login ID: page 1
Aux Work Reason Code Type
Use this field to specify whether the agent must enter a reason code when changing to the AUX work mode.
Valid entry system none requested forced
Usage
The systemwide settings apply. The agent must not enter a reason code when changing to the AUX work mode. The agent can enter a reason code when changing to the AUX work mode. The agent must enter a reason code when changing to the AUX work mode.
Related links Aux Work Reason Code Type on page 345 Reason Codes on page 844
COR
Use this field to assign a class of restriction (COR) number to the agent login ID. You can enter a value from 0 to 995. The default value is 1.
Coverage Path
Use this field to specify the coverage path used by the calls to the agent. The coverage path is used when:
· The agent is logged out. · The agent does not answer. · The agent is attending a personal call while logged in.
Valid entry blank 1 to 199
t1 to t199
Usage
Use this option if you do not want to enter a coverage path.
Use this option to enter a coverage path number. The upper limit might change depending on the system configuration. For more information, see Avaya Aura® Communication Manager System Capacities Table.
Use this option to enter a time-of-day table number.
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Agent Login ID
Direct Agents Calls First
Use this field to enable the system to select the direct agent calls before other ACD calls. The system displays this field only if you set the Call Handling Preference field to percentallocation.
Related links Call Handling Preference on page 52
Forced Agent Logout Time
Use this field to specify the time at which the system must log out the agent.
Valid entry blank
00, 15, 30, or 45 01 to 23
Usage
Use this option if you do not want to specify the time at which the system must log out the agent.
Use this option to enter the time in minutes.
Use this option to enter the time in hours.
Related links Timezone Offset on page 567
Login ID
This field displays the login id for the agent.
LoginID for ISDN Display
Valid Entry y
n
Usage
The Agent LoginID CPN (Calling Party Number) and Name are included in ISDN messaging over network facilities.
Default. The calling party name and number are determined by the conditions administered for the trunk group.
Related links Name on page 50
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Agent Login ID: page 1
Logout Reason Code Type
Use this field to specify whether the agent must enter a reason code when logging out.
Valid entry system none requested forced
Usage The systemwide settings apply. The agent must not enter a reason code when logging out. The agent can enter a reason code when logging out. The agent must enter a reason code when logging out.
Related links Logout Reason Code Type on page 345 Reason Codes on page 844
LWC Reception
Use this field to specify the location where the system must store the LWC messages.
Valid entry spe none audix
Usage Use this option to store the LWC messages on Switch Processor Element (SPE). Use this option if you do not want to store the LWC messages. Use this option to store the LWC messages on the voice messaging system.
Related links AUDIX Name on page 411
Maximum time agent in ACW before logout (sec)
Use this option to enter the maximum time an agent can be in ACW on a per agent basis.
Valid entry system none 30 to 9999
Usage
With this option, system-wide settings apply. This is the default option.
With this option, ACW timeout is inapplicable to the agent.
With this option, you can provide the number of seconds in a specific timeout period. The setting takes precedence over the system setting for maximum time in ACW.
Related links Maximum Time Agent in ACW before Logout (sec.) on page 348
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Agent Login ID
Messaging Server Name for Messaging
Use this field to assign a name for the messaging system that is used for LWC reception and that provides coverage for an agent login id. By default, the field is blank.
MIA Across Skills
Use this field to specify if the system must remove an agent from the MIA queue.
Valid entry system y
n
Usage
With this option, system-wide settings apply. This is the default option.
With this option, the system removes an agent from the MIA queues for all the splits or skills for which an agent is available when the agent answers a call from any assigned splits or skills.
With this option, the system removes ACW agents from the queue.
Related links MIA Across Splits or Skills on page 337
MWI Served User Type
Controls the auditing or interrogation of a served user's message waiting indicator (MWI).
Valid Entries fp-mwi qsig-mwi sip-adjunct blank
Usage The station is a served user of an fp-mwi message center. The station is a served user of a qsig-mwi message center. Used to audit message waiting lamps. The served user's MWI is not audited or if the user is not a served user of either an fp-mwi or qsig-mwi message center.
Name
Use this field to enter the name of the agent. You can enter up to 27 alphanumeric characters.
Note:
Supported by Unicode language display for the 4610SW, 4620SW, 4621SW, and 4622SW, 1600-Series, 9600-Series IP Deskphones, 9600-Series H.323 Deskphones, and Avaya J100 Series IP Phones. Unicode is also an option for the 2420J telephone when the Display Character Set is katakana. For more information about 2420J, see 2420 Digital Telephone User's Guide.
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Related links Display Character Set on page 821
Agent Login ID: page 1
Password
Use this field to assign a password to the agent login ID.
The Minimum Agent-LoginID Password Length field specifies a minimum length for the password.
Related links Minimum Agent-LoginID Password Length on page 330
Password (enter again)
Use this field to enter the same password that you entered for the Password field.
Port Extension
Use this field to assign an extension to the port used for AAS or voice messaging. You cannot use a vector directory number (VDN) or an agent login ID as a port extension.
Security Code
Use this field to enter the four-digit station security code for the Demand Print messages feature.
TN
Use this field to assign a tenant number (TN) to the agent login ID. The default value is 1. The upper limit might change depending on the system configuration. For more information, see Avaya Aura® Communication Manager System Capacities Table.
Unicode Name
This read-only field shows the assignment of Unicode Name to the agent.
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Agent Login ID
Valid entry n y
Usage Unicode Name is not assigned to the agent. Unicode Name is assigned to the agent.
For more information on Unicode Language Display, see Administering Avaya Aura® Communication Manager.
Agent Login ID: page 2
Call Handling Preference
Use this field to specify the call that the agent must receive when calls are in queue.
Valid entry
skill-level
greatest-need
percentallocation
Usage For the oldest high priority call waiting for an agent with the highest skill level. For the oldest high priority call waiting for an agent with any skill level. For the call waiting for an agent with a skill that deviates the most from the administered allocation.
This option is available only with Avaya Business Advocate.
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Agent Login ID: page 2
Direct Agent Skill
Use this field to specify the skill number used to handle direct agent calls. You can enter a value from 1 to 99. The upper limit might change depending on the system configuration. For more information, see Avaya Aura® Communication Manager System Capacities Table.
Local Call Preference
Use this field to enable the Local Call Preference feature. The system displays this field only if you set the Multiple Locations field to y. For calls queued in more than one skill for a multiskilled EAS agent, the system gives preference to the calls whose trunk location number matches the location number of the previously busy agent. This feature is available only with Call Center Release 3.0 and later.
Related links Call Center Release on page 843 Multiple Locations on page 838
PA
Use this field to allocate a percentage of the agent time for each skill that you assign to the agent. The total allocation for all agent skills must be equal to 100%.
RL
Use this field to administer the agent as a reserve agent for a skill or a group of skills. You can assign a reserve level or an interruptible level.
The system displays this field only if you enable the Business Advocate feature.
Valid entry 1 or 2 a m n blank
Usage Use this option to assign a reserve level to the agent. Use this option to assign the auto-in-interrupt interruptible level to the agent. Use this option to assign the manual-in-interrupt interruptible level to the agent. Use this option to assign the notify-interrupt interruptible level to the agent. Use this option if you do not want to assign a reserve level or an interruptible level to the agent.
Related links Interruptible Aux Threshold on page 409
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Agent Login ID
Service Objective
Use this field to enable the Service Objective feature. The agent receives calls according to the ratio of Predicted Wait Time (PWT) or Current Wait Time (CWT) and administration of the Service Objective feature. The Service Objective feature is available only with Avaya Business Advocate. The system displays this field only when you set the Call Handling Preference field to greatest-need or skill-level. Related links Call Handling Preference on page 52
SL
Use this field to specify a skill level for every skill that you assign to the agent. If you disable the Expert Agent Selection-Preference Handling Distribution (EAS-PHD) feature, you can enter up to two priority levels. If you enable the Expert Agent SelectionPreference Handling Distribution (EAS-PHD) feature, you can enter up to 16 priority levels.
SN
Use this field to assign a skill number to the agent. If you disable the Expert Agent Selection-Preference Handling Distribution (EAS-PHD) feature, you can enter up to four skills. If you enable the Expert Agent Selection-Preference Handling Distribution (EAS-PHD) feature, you can enter up to 20 or 60 skills depending on the system configuration.
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Chapter 8: Alias Station
Use the Alias Station screen to: · Administer new telephone types that are not supported by the system. · Map new telephone types to a telephone type that supports Alias Station. · Assign names to telephone types.
To change the value of the fields on the Alias Station screen, use the following command: change alias station
Alias Set Type
Use this field to assign a name to the nonsupported telephone type. You can enter up to five alphanumeric characters.
Supported Set Type
Use this field to enter a supported telephone type. For example, DCP, H.323, SIP, or analog telephone types.
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Chapter 9: Alphanumeric Dialing Table
Use the Alphanumeric Dialing Table screen to associate Alpha-name to digit strings. Users can make a data call by entering the Alpha-name instead of the digit string.
To change the value of the fields on the Alphanumeric Dialing Table screen, use the following command:
change alphanumeric-dial-table
Alpha-name
Use this field to assign a unique name to the digit string. The name must begin with an alphabet and can have up to eight alphanumeric characters.
Mapped String
Use this field to enter a string that you must use to generate the dialing string. You can enter up to 24 characters. You can also use the facility access codes in the string.
Valid entry
Usage
0 to 9
To enter a digit.
A to Z, a to To enter an alphabet. The system maps the uppercase alphabets to the lowercase
z
alphabets.
(, ), /, -, ^, To enter a special character. space
+
To enable the system to wait for a dial tone.
%
To indicate that rest of the digits are for end-to-end signaling.
,
To enable the system to pause for 1.5 seconds.
#
To enter a DTMF digit pound.
*
To enter a DTMF digit asterisk.
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Chapter 10: Announcements/Audio Sources
Use this screen to assign announcements to circuit packs and port locations. To change the value of the fields on the Announcements/Audio Sources screen, use the following command: add announcement n, where n is the extension number. In Communication Manager 7.1.3, you can configure up to 9000 announcements for a single Avaya Aura Media Server instance.
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Announcements/Audio Sources
Annc Name
Use this field to assign a name to the announcement. You can enter up to 27 characters. Announcement names on a single VAL circuit pack must be unique. You can enter alphabets, digits, and the following special characters: /, :, *, ?, <, >, \\, @, and space. The system saves the value of this field as the filename of the announcement.
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Annc Type
Use this field to specify the type of announcement.
Valid entry analog analog-m analog-fd aux-trunk aux-trk-m ds1-fd
ds1-ops ds1-sa
integrated integ-mus integ-rep
Usage
To play the announcement from an external device for a specific period and then disconnect. The caller hears a click when the device is disconnected.
To play the announcement continuously from an external announcement device.
To play the announcement from an external device for a specific period and then disconnect. The caller hears a click when the device is disconnected.
To play the announcement from an external announcement device with a 4-wire aux interface.
To play the announcement from an external announcement device that does not indicate that the playback is active.
To play the announcement from a DS1 port on the circuit pack. The caller does not hear a click when the device is disconnected. The system sends a disconnect message to the port to stop the playback when the announcement is complete.
To play the announcement from a DS1 port on the circuit pack. The caller does not hear when the device is disconnected.
To play the announcement from a DS1 port on the circuit pack. The caller does not hear a click when the device is disconnected. The system sends a disconnect message to the port to stop the playback when the announcement is complete.
To play the announcement stored internally on a special integrated announcement circuit pack. Use this option for general and VDN announcements.
To play the announcement from an integrated music source, or to play music stream.
To play the announcement with integrated repeating.
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COR
COR
Use this field to assign a class of restriction (COR) number to the announcement. You can enter a value from 0 to 995.
Extension
This field displays the extension number of the announcement. Note: If you enter a shortened extension from the multilocation dial plan while adding the announcement, then the system does not perform certain administration and validation tasks. Therefore, the system might not display the resultant warnings or submittal denials.
Group/Port/Media Server
Use this field to assign a board location or an audio group number to the announcement. For an integrated announcement, integ-mus and integ-rep announcements type, the system displays the field name as Source. You must enter the value by using one of the following formats:
· Gnn, where nn is an audio group number. · aaxss, where aa is a cabinet number, x is a carrier number, and ss is a slot number. · gggv9, where ggg is a gateway number. · Mxx, where xx is a media server number. Note: To administer DID Intercept announcements in a multilocation system where each location requires a different announcement, you must enter an audio group number instead of a VAL port number.
Live Stream Source
With Live Stream source, you can source music from external sources. Live Stream Source is visible only when Annc-type is set to integ-rep or integ-mus.
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Announcements/Audio Sources
Live Stream Source allows CM to use an Avaya Aura® Media Server (MS) stream source as an annc for music or as a repeating annc. Additional configuration is needed on AMS to define the stream source.
Valid Entry y
n
Usage
Music on hold is streamed live from an external source.
Live Stream Source can be set to y, only if Queue is set to b and the Source is set to media-server or audio group having all the members as mediaserver only. If Live Stream Source is set to y then the Protected field becomes Y and non-editable.
Identifies that music on hold is not streamed live from external source.
The default value is n.
Note:
Avaya Aura® Media Server (MS) streams music from an external source, using the concept of a StreamSource. A StreamSource is designed to simulate a music source such as a CD for a natural music-on-hold experience. Live StreamSource provides a radio broadcast effect and enables multiple endpoints to listen same audio stream, that plays continuously in the background.
Protected
Use this field to set the protection mode for an integrated announcement.
Valid entry Usage
y
A user cannot change or delete the integrated announcement using a telephone session or
an FTP session.
When you record or transfer an announcement file by using FTP, the system sets the announcement file to read only.
n
A user can change or delete the integrated announcement using a telephone session or an
FTP session.
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Queue
Valid Entry y
n b
Queue
Usage
Queues calls for the announcement if the announcement type is integrated, integrep or aux-trunk. The caller is always connected to the beginning of the announcement. ACD, vectoring delay announcements and call centers should always use this option. This is the default.
No queue nor barge-in. The caller is always connected to the beginning of the announcement. The announcement does not play if a port is unavailable.
Sets up barge-in if the announcement type is integrated, integ-rep or aux-trunk. When the announcement type is integ-mus, this field defaults to b. Callers are connected to the announcement at any time while it is playing.
Note:
The same non-barge-in announcement can be played through more than one port (or all ports) of an integrated circuit pack. The initial request to play an announcement selects an available port on the board on which the announcement resides. If there are additional requests to play the announcement while it is playing on another port(s), another port is selected. If all ports are busy, new requests to play announcements go to the integrated announcement system queue. Otherwise, the request to play is denied, and processing continues without the caller hearing the announcement. When a port becomes available, all queued calls (up to the platform "calls connected" limit) are connected at the same time to hear the announcement play from the beginning.
A barge-in announcement starts playing when first requested and continues playing through a port, repeating until there are no more requests. Call processing simultaneously connects calls to the playing barge-in announcement. Each call remains connected until the requesting feature operation removes the call (for example, wait step times out). Barge-in type announcements never select another port to play the same announcement once it is playing on a specific port.
Queue Length
Use this field to specify the number of calls that can queue for the announcement. The maximum allowed entry depends on the system configuration.
Rate
The recording rate speed in 1000 bits/seconds for TN750 or ISSPA integrated announcements. A different recording speed can be used for each integrated announcement. With VAL type sources, the default is 64 and cannot be changed.
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Announcements/Audio Sources
Valid Entry 16
32 64
Usage 16 kbps · 8 minutes and 32 seconds of announcement time per circuit pack · 1 hour and 24 minutes for 10 circuit packs for the TN750 · 240 minutes of storage time for the ISSPA This rate does not provide a high-quality recording. Do not use this rate for customer announcements, but it is adequate for VDN of Origin announcements. 32 kbps · 4 minutes and 16 seconds of total announcement time for the TN750 · 120 minutes of storage time for the ISSPA 64 kbps · 2 minutes and 8 seconds of announcement time per circuit pack · 42 minutes for 10 circuit packs for the TN750 · 60 minutes of storage time for the ISSPA This is the default for VAL
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
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Chapter 11: ARS Toll Table
Use this screen to assign ARS toll tables used by Subnet Trunking. This screen specifies whether calls to the local central office codes listed on the table are toll calls. To change the value of the fields on the ARS Toll Table screen, use the following command: change ars toll
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Chapter 12: Attendant Console
Use this screen to assign an attendant console to the system. To add an attendant, use the following command: add attendant n, where n is the attendant console number.
Attendant console: page 1
Attendant Console x
Number assigned to the attendant console. The attendant console is the main answering position for an organization.
Auto Answer
Use this field to administer the setting of the auto answer button on the attendant console.
Valid entry all acd none
Usage Enables automatic answering for all incoming calls. Enables automatic answering only for ACD and direct agent calls. Disables automatic answering.
Console Type
Use this field to specify the type of attendant console.
You can administer only one night-only or day/night type console in the system unless the Tenant Partitioning feature is enabled.
Night Service can be activated from the principal attendant console or the station on which the nite-serv button is administered.
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Attendant console: page 1
Valid entry principal day-only night-only day/night
Usage To enable the attendant console to activate Night Service. To enable the attendant console to handle only day service calls. To enable the attendant console to handle only night service calls. To enable the attendant console to handle day or night service calls.
COR
Use this field to assign a class of restriction (COR) number to the attendant console. You must enter a value from 0 to 995.
COS
Use this field to assign a class of service (COS) number to the attendant console. You must enter a value from 0 to 15.
Data Module
Use this field to enable the connection between the attendant console and the data terminal. Related links
Data Module on page 190
Disp Client Redir
Use this field to specify whether the attendant console must display the redirection information for a call originating from the station with the Client Room Class of Service feature.
Note:
You must set the Disp Client Redir field to y for:
· The stations with the audix type. · The stations with the audix voice power ports or the ports that require the display
information.
The system displays this field only if the Hospitality feature is enabled.
Related links Hospitality (Basic) on page 834 Hospitality (G3V3 Enhancements) on page 835
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Attendant Console
Display Language
Use this field to specify the language in which the system must display the console messages. You must enter one of the following values:
· english · french · italian · spanish · user-defined · unicode Note: The unicode option is available only for unicode-supported telephones, such as 4610SW, 4620SW, 4621SW, 4622SW, 2420J, Sage, Spice, Spark, 9600-series telephones, and Avaya J100 Series IP Phones. Related links Display Character Set on page 821
Extension
Use this field to assign an extension to the attendant console. Users can call the attendant console by dialing this extension. If an attendant console does not have an extension, the attendant console can only be addressed as a member of the attendant group. You must not leave this field blank for the attendant console with a data module.
Group
Use this field to assign an attendant group number to the attendant console.
H.320 Conversion
Use this field to enable the conversion of H.320-compliant calls to voice-only calls for the attendant console.
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Attendant console: page 1
Note: The system can handle only a limited number of conversion calls. Therefore, the number of attendant consoles with H.320 conversion must be limited.
Name
Use this field to assign a name to the attendant console. You can enter up to 27 alphanumeric characters.
Note:
Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Port
Use this field to assign a port number to the attendant console. For reliability, the attendant consoles must not be assigned to ports on the same digital line circuit pack. You must enter one of the following values:
· A port number. · ip for a soft attendant console. · X if no hardware is associated with the port assignment. You can assign the port number using one of the following formats: · cabinet number : carrier number : slot number : circuit number
Valid entry 01 to 64 A to E 01 to 20 01 to 31
Usage The first and second characters represent the cabinet number. The third character represents the carrier number. The fourth and fifth characters represent the slot number. The sixth and seventh characters represent the circuit number.
· gateway number : module number : circuit number
Valid entry 001 to 250 V1 to V9 01 to 31
Usage The first three characters represent the gateway number. The fourth and fifth characters represent the module number. The sixth and seventh characters represent the circuit number.
Related links IP Attendant Consoles on page 835 Extension on page 66
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Attendant Console
Security Code
The station security code required by the SoftConsole IP attendant.
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
Type
Valid Entry console 302
Usage The extension is used with any type of console. 302B/C/D or SoftConsole IP attendant
Direct Trunk Group Select Button Assignments (Trunk Access Codes)
Use this field to assign the local and remote trunk access codes (TACs) to the attendant console. A local TAC refers to a trunk group or a Loudspeaker Paging zone on the local server. A remote TAC refers to a trunk group on the remote server.
The local TAC can have up to four digits and the remote TAC can have up to three digits. You can use the asterisk (*) and the pound sign (#) as the first digit.
Note:
The local TAC between the local and remote servers must be a DCS trunk. If not, Communication Manager cannot monitor the remote trunk.
Hundreds Select Button Assignments
Use this field to assign a hundreds group to a Hundreds Select button on an attendant console. You can enter up to 11 digits.
Use 1 to 8 fields for 24A consoles, and use 1 to 8 fields for 26A consoles.
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Attendant console: page 2 Softconsole IP Attendant
Attendant console: page 2 Softconsole IP Attendant
Always Use
Use this field to enable a soft attendant console to register, regardless of the emergency call handling settings on the console. If the user dials 911, the system uses the administered value in the Emergency Location Ext field. The Always Use field is inapplicable to SCCAN wireless consoles and the consoles administered as the H.323 type. Related links Emergency Location Ext on page 770
Direct IP-IP Audio Connections
Use this field to enable direct audio connections between IP endpoints. Direct audio connections save bandwidth resources and improve the sound quality of voice over IP transmissions.
Note: This field has no impact on a SIP station (OPS) on shuffling. For SIP stations, this field is controlled by the "Direct IP-IP Audio Connections" field in the Signaling group form.
Emergency Location Ext
Use this field to specify the emergency location extension for the soft attendant console. You can enter up to 16 digits. This extension is used as a starting point for identifying the location when an emergency call is made. The system uses:
· The CAMA Numbering format to modify the extension before sending the extension over the CAMA trunk.
· The Numbering-Public/Unknown format to modify the extension before sending the extension over the ISDN trunk.
IP Audio Hairpinning
If enabled, IP endpoints connect through the IP circuit pack in the server in IP format to bypass the Communication Manager TDM bus.
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Attendant Console
Remote Softphone Emergency Calls
Use this field to specify how Communication Manager must handle emergency calls from an IP telephone. The system displays this field only when IP Softphone is enabled for the system.
Valid entry
Usage
as-on-local If the administrator populates the IP Address Mapping screen with emergency numbers, the value as-on-local functions as follows:
· If the Emergency Location Extension field in the Station screen is the same as the Emergency Location Extension field in the IP Address Mapping screen, the feature sends the console's own extension number to the Public Safety Answering Point (PSAP).
· If the Emergency Location Extension field is different from the Emergency Location Extension field in the IP Address Mapping screen, the value as-on-local sends the extension in the IP Address Mapping screen to the Public Safety Answering Point (PSAP).
block
Prevents the completion of emergency calls. Used for users who are mobile but always have a circuit-switched telephone nearby, and for those who are farther away from the media server or switch than an adjacent area code served by the same 911 Tandem office. When users attempt to dial an emergency call from an IP Telephone and the call is blocked, they can dial 911 from a nearby circuit-switched telephone instead.
cesid
Communication Manager can send the CESID information supplied by the IP Softphone to the PSAP. The end user enters the emergency information into the IP Softphone.
Used for IP Softphones with road warrior service that are near enough to the Avaya S8XXX server that an emergency call routed over the trunk reaches the PSAP that covers the server or switch.
If the Avaya S8XXX server uses ISDN trunks for emergency calls, the digit string is the telephone number, provided that the number is a local direct-dial number with the local area code, at the physical location of the IP Softphone. If the Avaya S8XXX server uses CAMA trunks for emergency calls, the end user enters a specific digit string for each IP Softphone location, based on advice from the local emergency response personnel.
option
Users can select the option (extension, block, or cesid) that the user selected during registration and the IP Softphone reported. Used for extensions that can be swapped back and forth between IP Softphones and a telephone with a fixed location.
Users choose between block and cesid on the softphone. A DCP or IP telephone in the office automatically selects extension.
Related links IP Softphone on page 745
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Attendant console: page 2 VIS feature options
Service Link Mode
Use this field to specify the duration of a service link connection. The service link is the combined hardware and software multimedia connection between an H.320 Desktop Video Conferencing (DVC) system and Communication Manager.
The service link is established when a user receives or makes a call during a multimedia, IP softphone, or IP telephone session.
Valid entry asneeded
perman ent
Usage
For multimedia, IP softphone, and IP telephone users. The service link remains connected for 10 seconds after the user disconnects a call so that the user can immediately make or receive another call. After 10 seconds, the link is disconnected, and a new link must be established to make or receive a call.
For call center agents who are constantly making or receiving calls during the multimedia, IP softphone, or IP telephone session. The service link remains connected for the entire duration of the session.
Attendant console: page 2 VIS feature options
VIS FEATURE OPTIONS
These fields administer Visually Impaired Services options.
Auto Start
Use this field to allow or deny permission to an attendant to start a call by pressing any key on the keypad without the need to first press the Start button.
Echo Digits Dialed
Enables and disables voiced confirmation of dialed digits.
Attendant console: page 3
Displays if the attendant console is to be connected to a data terminal using a 7400B or 8400 data module.
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Attendant Console
Related links Data Module on page 65
ATTENDANT DATA MODULE
Bcc
Determines compatibility when non-ISDN facilities are connected to ISDN facilities (ISDN Interworking feature).
COR
Valid Entry 0 to 995
Usage The Class Of Restriction (COR) number for the data module.
COS
Valid Entry 0 to 15
Usage The (COS) number used to designate allowed features.
Data Extension
Use this field to assign an extension to the data module. You can enter up to five digits. This value must conform to the dial plan.
Name
The name assigned to the data module extension number. Related links
Ext on page 74
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
ABBREVIATED DIALING
List1
Valid Entry s g
Usage System Group -- A group number is also required
Table continues...
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Valid Entry p e
Usage Personal -- A personal list number also is required Enhanced
Attendant console: page 3
SPECIAL DIALING OPTION
Use this field to specify the destination of a call when the data module originates the call. Use this field to specify the destination of a call that starts from the data module.
Valid entry hot-line
default
Usage
Single-line telephone users can automatically make a call to an extension, a telephone number, or a feature access code.
The system dials an Abbreviated Dialing number when the user disconnects.
HOT LINE DESTINATION -- Abbreviated Dialing Dial Code
Abbreviated Dialing Dial Code is the Abbreviated Dialing number dialed when the user goes offhook on a Data Hotline call.
Using Hotline Service, IP telephone users can place a call to a preassigned destination by going offhook. The preassigned destination can be an extension, a telephone number, or a feature access code. The Hotline Service feature does not work for SIP endpoints.
The Hotline Service destination number is stored in Abbreviated Dialing List.
A Hotline Service destination can be a Direct Department Calling (DDC), a Uniform Call Distribution (UCD), a Terminating Extension Group (TEG) extension, or any individual extension within a group. Also, any extension within a DDC group, UDC group, or TEG can have Hot Line Service assigned.
Use Hotline Service when you require very fast service and when you use a telephone only for gaining access to a certain facility. You can use Loudspeaker Paging Access with Hotline Service to provide automatic access to paging equipment.
This field is available only for a default or hotline special dialing option.
Related links SPECIAL DIALING OPTION on page 73
DEFAULT DIALING Abbreviated Dialing Dial Code
The AD number dialed when the user goes off-hook and enters a carriage return following the "DIAL" prompt. The data call originator also can perform data terminal dialing by specifying a dial
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Attendant Console
string that might or might not contain alphanumeric names. Available only for a default special dialing option. Related links SPECIAL DIALING OPTION on page 73
ASSIGNED MEMBER
Ext
The extension number of a previously-administered user who has an associated Data Extension button and who shares the use of the module.
Name
The name assigned to the data module extension number. Related links
Ext on page 74
Attendant console: page 3 feature button assignments
FEATURE BUTTON ASSIGNMENTS
Administers the feature buttons assigned to the attendant console. The split and forced release buttons are administered in a fixed location. The hold, night-serv, and pos-busy buttons have default locations. The following table provides descriptions of feature buttons that are unique to the attendant console.
Audible Tones On/Off
Valid Entry cw-ringoff
in-ringoff
re-ringoff
Usage
Call waiting ringer off; turns on/off the audible tone for call waiting on attendant console (1 per console).
Incoming call ringer off; turns on/off the audible tone for incoming call ringer (1 per console).
Timed reminder ringer off; turns on/off the audible tone for timer reminder ringer (1 per console).
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Attendant console: page 3 feature button assignments
Attendant Control of Trunk Group Access
Valid Entry act-tr-grp deact-tr-g class-rstr em-acc-att hold pos-busy
serial-cal override intrusion dont-split vis
Usage
Activate trunk group access. The attendant can control a trunk group. All calls going to the trunks are routed to the attendant (one per console).
Deactivate trunk group access. The attendant can release control of a trunk group (one per console).
Display Class of Restriction. Used to display the COR associated with a call (one per console).
Emergency Access to the Attendant. The associated status lamp is flashed when there are one or more calls on the emergency attendant queue (one per console).
Hold. When the Hold button is pressed while the attendant is active on a loop, the party on the loop is put on hold and the call type button associated with the loop is lit (one per console).
Position Busy. When this button is pushed, the attendant is put into position busy mode, the "Pos Avail" light is turned off, and the light associated with the pos-busy button is lit. Pushing the pos-busy button a second time takes the console out of "position busy" mode, turns on the "Pos Avail" light and turns off the light associated with the pos-busy button.
If the pos-busy button is administered on a 2-LED button, the top LED flashes when the last attendant goes into "Position Busy" mode. Otherwise, if the button has only one LED, the single LED associated with the pos-busy button flashes (one per console).
Serial Call. The attendant-extended calls can return to the same attendant if the trunk remains off-hook (one per console).
Attendant Override. The attendant can override diversion features such as, Call Forwarding, Call Coverage, and so on (one per console).
Call Offer. The attendant can extend a call when the called party is active on another call (one per console).
Don't Split. The attendant cannot split away a call when dialing (one per console).
Visually Impaired Attendant Service (vis). Activates visually impaired service for the attendant. When this service is activated, the attendant can listen to console status or messages by pressing buttons that have been translated as follows:
· "con-stat" repeats the console status.
· "display" calls out display contents.
· "dtgs-stat" calls out the DTGS status.
· "last-mess" repeats the last message.
· "last-op" calls out the last operation.
Trunk Group Select Up to 12 DTGS buttons can be administered. The status lamp associated with the feature button is used to monitor the busy/idle status of the trunk. Trunk groups administered on these buttons cannot be controlled using Attendant Control of Trunk Group Select buttons.
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Attendant Console Valid Entry local-tgs remote-tgs
Other
Valid Entry alt-frl hundrd-sel
group-disp group-sel occ-rooms maid-stat vu-display
Usage
Local trunk group select. The attendant can access trunk groups on the local server running Communication Manager.
Remote trunk group select. The attendant can access trunk groups on a remote server running Communication Manager.
Usage
Alternate facility restriction level. The attendant can activate or deactivate the AFRL feature. When activated, the originating device (lines or trunks) can use an alternate set of the facility restriction levels to originate a call (one per console).
Hundreds group select. Additional administered hundreds group select feature buttons. When a feature button is administered as "hundrd-sel", the system displays a subfield that must be administered with a 1 to 3 digit hundreds group plus prefix, if needed. Administered hundrd-sel feature buttons operate in the same manner as fixed HGS buttons.
The total number of hundreds group select buttons (fixed and administered) allowed on a console is 20. Therefore, if all 20 fixed HGS buttons have been administered, no additional hundrd-sel feature buttons can be administered.
If 12 HGS buttons are assigned, reassign the night, pos-busy, and hold buttons to locations 20, 21, and 3, respectively. The HGS buttons should then be assigned to the right-most three columns, as required.
Group Display. The attendant can see a display of extensions currently being tracked on the DXS module.
Group Select. The attendant can select a specific group of hundreds by dialing the first 2 or 3 digits of the hundreds group.
Occupied rooms. The attendant can see which rooms are occupied.
Maid status. The attendant can see which rooms are in one of six specified states.
VuStats (vu-display). Users with display telephones and attendants can turn on the VuStats display. The limit to the number of VuStats feature buttons depends on how many feature buttons are available on the attendant console. You can set up a separate VuStats display format for each feature button. Therefore, agents can change the type of measurements on their display by selecting a different VuStats feature button.
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Attendant console: page 4 display module button assignments
Attendant console: page 4 display module button assignments
Display Module Button Assignments
Display-type buttons obtain display functions on the associated alphanumeric display. Also, several feature buttons can be administered so that their associated status lamps can be used to provide visual indications of the associated feature or function. In some cases, the button itself is inoperative. These buttons are noted as [status lamp]. If a Call Cover Msg Rt (cov-msg-rt) button is assigned, a Leave Word Calling Delete Msg (delete-msg) button and a Next (next) button must also be assigned.
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Chapter 13: Audio Group
Use this screen to assign an audio group to the system. The audio group is a collection of recorded audio sources that have been placed in the group to facilitate the selection. To add an audio group, use the following command: add audio-group n, where n is the audio group number.
Audio Source Location
Use this field to assign a board location to the audio group. You must enter the value using one of the following formats:
· aaxss, where aa is a cabinet number, x is a carrier number, and ss is a slot number. · gggvx, where ggg is a gateway number and vx is the module number. · Mx, where x is the media server number.
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Group Name
Group Name
Use this field to assign an alphanumeric name to the audio group. Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
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Chapter 14: AUDIX-MSA Node Names
Example command: change node-names audix
AUDIX Names
The name of the voice messaging node consisting of a one- to seven-character string. Used as a label for the associated IP address. The node names must be unique on each server running Communication Manager.
IP Address
The IP address associated with the node name.
MSA Names
Identifies the name of the MSA node. Accepts a one- to seven-character string. The MSA names must be unique on each server running Communication Manager.
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Chapter 15: Authorization Code -- COR Mapping
Use this screen to administer an authorization code and the class of restriction (COR) number associated with the authorization code.
Security alert: To maximize security, you must:
· Administer authorization codes to the acceptable maximum length. · Create random authorization codes. · Change authorization codes at least quarterly. · Deactivate authorization codes immediately if a user leaves the company or changes
assignments. · Assign authorization codes with the minimum level of calling permission. To change the value of the fields on the Authorization Code -- COR Mapping screen, use the following command: change authorization-code n, where n is the authorization code.
AC
Use this field to administer an authorization code (AC). The AC can be any combination of numbers and can consist of 4 to 13 digits. To enhance the system security, you must administer the AC of 13 random digits. Related links Authorization Code Length on page 298
COR
Use this field to assign a class of restriction (COR) number to the authorization code. You can enter a value from 0 to 995.
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Authorization Code -- COR Mapping
Number of Codes Administered
The number of authorization codes already administered. There is a maximum number of authorization codes depending on system configuration.
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Chapter 16: Best Service Routing
Use this screen to enable the system to compare the specified skills, identify the skill that provides best service to a call, and deliver the call to an agent in the skill. If no agents are available in the skill, the call is queued. You can also use this screen to monitor the status of the specified skills and adjust call processing and routing as appropriate. To change the value of the fields on the Best Service Routing screen, use the following command: change best-service-routing n, where n is the routing number
Interflow VDN
Use this field to specify the dial access code that the user must use to access the interflow vector directory number (VDN) at the remote location. You can enter up to 16 alphanumeric characters.
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Best Service Routing
Valid entry 0 to 9, *, # ~p ~w or ~W ~m ~s
Usage To enter the digit. To enable the system to pause. To enable the system to wait. To enable the system to mark. To enable the system to suppress.
Location Name
Use this field to assign a name to the location where the Best Services Routing feature is enabled. You can enter up to 15 alphanumeric characters.
Lock
Use this field to provide additional security by preventing the system from sending any Best Service Routing (BSR) information to Call Management System.
TG
Valid entry
y
n
Usage
Does not send the BSR information to Call Management System. Sends the BSR information to Call Management System.
Maximum Suppression Time
Use this field to prevent callers from connecting to a VDN during the specified period after receiving a busy signal.
Valid entry 0 to 60
Usage The maximum poll suppression time in seconds.
Net Redir
Use this field to enable Network Call Redirection.
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Num
Num
Use this field to specify a location number. You can enter a value from 1 to 255.
Number
This field displays the current BSR plan number. The system displays a value from 1 to 255.
Status Poll VDN
Use this field to specify the dial access code that the user must use to access the Status Poll vector directory number (VDN) at the remote location. You can enter up to 16 alphanumeric characters.
Valid entry 0 to 9, *, # ~p ~w or ~W ~m ~s
Usage To enter the digit. To enable the system to pause. To enable the system to wait. To enable the system to mark. To enable the system to suppress.
Switch Node
Valid Entry 1 to 32767 blank
Usage
The Network Node ID for each switch if the Universal Call ID is being used. This is optional.
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Chapter 17: Bulletin board
Use this screen to provide the messages that the system must display when a user logs in. The first 10 lines are reserved for high-priority messages, which only an init or an inads user can enter. The system also displays the date on which the message was posted. To change the value of the fields on the Bulletin board screen, run the following command: change bulletin-board
Date
The date the bulletin board information was entered or last changed.
Text lines
Used for high priority messages on the bulletin board. Anyone with an init or inads login can enter high-priority information to trigger the high-priority message at login time. Additional lines can be used by anyone with access.
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Chapter 18: Button Type Customization Restriction
Use this screen to: · Prevent users from accessing the customized button labels. · Manage system allocation of the customized button labels.
To change the value of the fields on the Button Type Customization Restriction screen, run the following command:
change button restriction
Restrict Customization Of Button Types
Valid Entry y n
Usage
Restricts the use of customized feature button labels. This is the default.
Users can customize labels for all buttons on their telephones without any restrictions.
Restrict Customization Of Labels For the Following Button Types
The button type restricted from being customized.
Note: The abr-spchar and the abrv-dial button types require a special associated character.
· abrv-dial An abbreviated dialing list type: e = enhanced, g = group, p = personal, and s = system
· abr-spchar
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Button Type Customization Restriction
A ~-character in an abbreviated dialing digit string. ~, ~m, ~p, ~s, ~w, and ~W. For more information on digit strings, see Dial Code. Available only when Restrict Customization of Button Types is enabled. Related links Restrict Customization Of Button Types on page 87 DIAL CODE on page 35
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Chapter 19: Call Type Digit Analysis Table
Use this screen to specify how the system must modify telephone numbers when a user makes a call to an internal contact using the call log or a corporate directory. To change the value of the fields on the Call Type Digit Analysis Table screen, run the following command: change calltype analysis
Delete
Use this field to specify the number of digits that Communication Manager must delete from the starting of the dialed string. You can enter a value from 0 to 28.
Note: The value in the Delete field must not exceed the value in the Min field.
Dialed String length (Min, Max)
Communication Manager compares digit strings of this length to the original digit string, looking for a match to complete analysis and routing.
Dialed String Match
Use this field to specify the string that Communication Manager must use to match the dialed number for analysis and routing. The string can be up to 18 characters long and can consist of:
· Digits from 0 to 9 · The plus sign (+) · The wildcard characters, x and X
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Call Type Digit Analysis Table
Note: The plus sign (+), if used, must be the first digit in the string.
Insert
Communication Manager inserts these digits into the left-hand side of the original digit string to complete analysis and routing.
Location
Valid Entry Numeric value all
Usage
Telephones dialing from this location use the entries in the Call Type Digit Analysis Table. If there are matching entries in the telephone's location, those entries are used. If there are no matching entries in the telephone's location, the Communication Manager tries the entries in location all.
Type
Use this field to specify the call type associated with the dialed string.
Valid entry aar ars ext udp
Usage
For Automatic Alternate Routing (AAR) calls. For Automatic Route Selection (ARS) calls. For calls with extensions in the dial plan analysis table. For calls with extensions in the uniform dial plan table.
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Chapter 20: Call Vector
Use this screen to run a series of commands to specify how the system must handle calls directed to a vector directory number (VDN). For example, to change the value of the fields on the screen, run the following command: change vector n, where n is the VDN.
01 through XX
Use this field to enter the vector commands. The maximum number of commands that you can enter depends on the system configuration.
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Call Vector
Valid entry adjunct routing announcement busy check collect
consider
converse-on
disconnect
goto messaging
queue-to reply-best
return
route-to
set stop wait-time
Usage
To send a message to an adjunct for routing instructions.
To provide an announcement to the caller.
To provide a busy signal to the caller and to end the vector processing.
To check the status of a split or skill.
To collect the Caller Information Forwarding (CINFO) digits. The user can enter up to 16 digits.
To define the resource that the system selects as part of a Best Service Routing (BSR) consider series and to get the data that BSR uses to compare resources.
To direct a call to a split or skill and activate the voice response script in a Voice Response Unit (VRU).
To disconnect a call. The system plays an announcement before the call is disconnected.
To move the vector processing to an earlier or later step in the vector.
To enable the user to leave a message for the specified extension or the active VDN extension.
To queue a call to a split or skill and assign a priority level to the call.
To return the data to the primary vector on the server. Use this option only in status poll vectors and BSR applications.
To return the vector processing to the step following the goto command after a subroutine call is processed.
To route a call to the destination specified in the administered digit string.
Note:
Communication Manager prepends a plus sign (+) to the called number when Network Call Redirection is enabled.
To perform arithmetic and string operations and assign value to a vector variable.
To stop the processing of subsequent vector steps.
To delay the vector processing of the next step.
ANI/II-Digits
Indicates whether the ANI and II-Digits vector routing commands can be used or not. ANI/II-Digits Routing is a G3V4 Enhanced feature.
Related links G3V4 Enhanced on page 94
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ASAI Routing
ASAI Routing
Indicates whether or not the CallVisor Adjunct/Switch Applications Interface (ASAI) Routing option is active or inactive. Related links Attendant Vectoring on page 830
Attendant Vectoring
Indicates whether or not Attendant Vectoring is selected on this VDN. Attendant Vectoring does not support Call Center features. Related links Attendant Vectoring on page 830
Basic
Indicates whether Vectoring (Basic) is enabled for system. Related links
Vectoring (Basic) on page 845
BSR
Indicates if the Best Service Routing option is enabled on the system. BSR commands and command elements can be used in vectors only if this option is enabled. Related links Vectoring (Best Service Routing) on page 845
CINFO
Indicates whether Caller Information Forwarding (CINFO) Routing is enabled on the system. A call with CINFO routes based on the digits supplied by the network in an ISDN-PRI message. Related links Vectoring (CINFO) on page 846
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Call Vector
EAS
Indicates if Expert Agent Selection (EAS) is enabled on the system. Note: When EAS is enabled, the terminology changes from "Split" to "Skill" for help messages, error messages, and vector commands. For example, check backup split becomes check backup skill.
Related links Expert Agent Selection (EAS) on page 843
G3V4 Adv Route
Indicates whether G3V4 Advanced Vector Routing commands are allowed. Related links
Vectoring (G3V4 Advanced Routing) on page 846
G3V4 Enhanced
Indicates whether G3V4 Enhanced Vector Routing commands and features are allowed. Related links
Vectoring (G3V4 Enhanced) on page 846
Holidays
Indicates whether Holiday Vectoring features are allowed. Related links
Vectoring (Holidays) on page 846
LAI
Indicates if Look-Ahead Interflow is enabled.
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Lock
Related links Lookahead Interflow (LAI) on page 844
Lock
Controls access to the vector from the Call Management System.
Security alert: Always lock vectors that contain secure information (for example, access codes).
Valid Entry y
n
Usage
This vector is inaccessible from the CMS. Locked vectors can only appear and be administered through the SAT or a terminal emulator. If Meet-me Conference is enabled, Lock must also be enabled.
Gives CMS users the ability to administer this vector.
Meet-me Conf
Indicates if the Meet-me Conference feature is enabled. If enabled, designates the VDN as a Meet-me Conference VDN. Attendant Vectoring and Meet-me Conference cannot be enabled at the same time.
Related links Enhanced Conferencing on page 833
Multimedia
Valid Entry y
n
Usage
Multimedia calls are set to be answered and billed at the start of vector processing. Multimedia Call Handling must be enabled.
The vector stops receiving multimedia calls. Multimedia Call Handling is disabled. This is the default.
Related links Multimedia Call Handling (Basic) on page 837 Multimedia Call Handling (Enhanced) on page 837
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Call Vector
Name
Use this field to assign a name to the vector. You can enter up to 27 alphanumeric characters. The system displays this field only if you enable the Network Call Redirection feature. Related links ISDN/SIP Network Call Redirection on page 836 Display Character Set on page 821
Number
The vector number.
Prompting
Indicates whether Vectoring (Prompting) is enabled for the system. Related links
Vectoring (Prompting) on page 846
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Chapter 21: Calling Party Number Conversion for Tandem Calls
Use this screen to administer calling party number formats for tandem calls. Tandem calls are not originated locally on the switch. For tandem calls to the public network, the system might not provide the correct calling party information. To rectify this problem, Communication Manager changes the calling party number. Communication Manager compares the incoming calling party number with the calling party digit lengths, calling party prefixes, and trunk groups. When a match is found, Communication Manager creates the calling party number by deleting the digits specified in the Delete field, and then inserting the digits specified in the Insert field or the values of Numbering Plan Identifier (NPI) and Type of Number (TON) for Calling Party Number Communication Manager uses the numbering format specified in the Number Format field.
Note: Communication Manager does not update Calling Party Number in the NCA-TSC SETUP message. To map numbers for ISDN-SIP interworking, Communication Manager uses the values of the fields on the Calling Party Number Conversion for Tandem Calls screen or sends the call to Session Manager. Session Manager uses the number adaptation module for the mapping.
Note: During ISDN to SIP mapping, the number adaptation module does not have access to NPI and TON values in the incoming ISDN number. During SIP to ISDN mapping, Communication Manager uses the leading plus (+) sign to process an incoming SIP number as an international number and insert the plus (+) sign in the resulting number. To process an incoming SIP number, Communication Manager uses the values of the fields on the Calling Party Number Conversion for Tandem Calls screen. The following rules apply to an incoming calling party number with a leading plus (+) sign: · For an incoming SIP calling party number with a leading plus (+) sign, Communication Manager includes the plus (+) sign when searching for a match. · For an incoming SIP calling party number with a leading plus (+) sign, if a match is not found with the plus (+) sign, Communication Manager removes the plus (+) sign. · For an incoming ISDN calling party number reconstructed as an international number, Communication Manager searches for a match. If a match is not found, Communication Manager adds a leading plus (+) sign.
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Calling Party Number Conversion for Tandem Calls
For an incoming SIP calling party number, if Communication Manager finds a match, the matching entry overrides the setting of the Prepend `+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers field for the SIP signaling group. Use one of the following ways to modify calling party numbers for a SIP endpoint:
· Through the Calling Party Number Conversion for Tandem Calls form. · Through the System Parameters and Location Parameters forms. Note: Do not use both the options at the same time to modify the calling numbers sent to SIP endpoints. Using both the options at the same time leads to inconsistent display. For example, the calling number display on a direct incoming call may differ from the display if that same incoming call is forwarded to another station. If you choose the System Parameters and Location Parameters forms, you can use the Calling Party Number Conversion for Tandem Calls form for other purposes. To do so, use the Outgoing Trunk Group (s) column to ensure that no entry on that form affects the trunk groups used for SIP stations. To modify the value of the fields on the Calling Party Number Conversion for Tandem Calls screen, use the following command: change tandem-calling-party-num
Related links Delete on page 99 Insert on page 99 Incoming Number Format on page 100
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Modify Tandem Calling Number on page 527
CPN Prefix
CPN Prefix
Valid Entry 0 to 9, +
any
Usage
Any combination of digits 0 to 9, up to 16 digits to indicate the calling party prefix. The '+' sign is also valid.
Any incoming calling number matches the entry.
Delete
Valid Entry 1 to 15 all blank
Usage
The number of digits deleted when modifying the tandem calling party number.
All digits are deleted.
No digits are deleted from the received calling party number. This is the default value.
Insert
Valid Entry 0 to 9, + natl-prefix
intl-prefix
blank
Usage
Any combination of digits 0 to 9, up to 16 digits. The '+' sign is also valid.
Communication Manager adds the Long distance access code number as the prefix to the calling-party number. The Long distance access code field is on page 1 of the Location parameter screen. If the Long distance access code field is blank, Communication Manager uses the prefix number in the National CPN Prefix field on the Feature related system parameter screen.
Communication Manager adds the International access code number as the prefix to the calling-party number. The International access code field is on page 1 of the Location parameter screen. If the International access code field is blank, Communication Manager uses the prefix number in the International CPN Prefix field on the Feature related system parameter screen.
The received calling party number is not prefixed with any digits. This is the default value.
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Calling Party Number Conversion for Tandem Calls
Incoming Number Format
Use this field to specify the numbering format of the calling party number.
Note:
Unlike ISDN calling party numbers, SIP calling party numbers do not include a numbering format. The only descriptor is the optional leading plus sign (+). Therefore, to match the calling number in an incoming SIP trunk call, you must do one of the following:
· To match a calling number that does not start with a plus sign (+), you must leave the Incoming Number Format field blank.
· To match a calling number that starts with a plus sign (+), you can leave the Incoming Number Format field blank, or you can set the field to intl-pub.
Valid Entry intl-pub, lev0-pvt, lev1-pvt, lev2-pvt, locl-pub, natl-pub, pub-unk, unk-unk blank
Usage The numbering format to use in modifying the tandem calling party number.
All numbering formats match this entry.
Len
Valid entry 1 to 16 any
Usage
A number between 1 and 16 that indicates the length of calling party number.
A calling party number of any length.
Note:
If after insertion and deletion the resulting calling number is longer than 21 digits, the system will not perform the conversion.
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Outgoing Number Format
Outgoing Number Format
Valid Entry intl-pub, lev0-pvt, lev1-pvt, lev2-pvt, locl-pub, natl-pub, pub-unk, unk-unk blank
Usage The numbering format to use in modifying the tandem calling party number.
All numbering formats match this entry.
Outgoing Trunk Group(s)
Valid entry
Trunk group number
blank
Usage Enter an ISDN or a SIP outgoing trunk group number.
If you leave this field blank, any ISDN and SIP trunk group matches this entry, if it has the Modify Tandem Calling Number field enabled. By default, this field is blank.
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Chapter 22: CAMA Numbering Format
Administers the Centralized Automatic Message Accounting (CAMA) trunks and provides Caller's Emergency Service Identification (CESID) information to the local community's Enhanced 911 system through the local tandem office. Provides the CESID format by extension number or number blocks. Multiple CESID formats can be sent over multiple CAMA trunk groups allowing for mixed station numbering plans and some limited conversion from non-DID to DID numbers typically required by the Private Switch/Automatic Location Interface (PS/ALI) database. Example command: change cama-numbering
CESID
The number used to identify the calling terminal within an emergency service system. This field can represent a prefix to an extension or the entire CESID. Accepts up to 16 digits. The maximum number of allowed digits depends on the equipment.
Ext Code
The leading digits or all of the digits in the extension for the specified CESID. If the extension length is greater than the number of digits in the extension code, the extension code is interpreted as a block of digits. Example: If the extension length is 4 and the extension code is 11, the CESID serves extensions 1100 through 1199.
Ext Len
The number of digits in the extension. Accepts up to 16 digits.
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System CESID Default
System CESID Default
The number sent over the CAMA trunk if the Extension Code for the CESID is unspecified. Accepts up to 16 digits. Related links Ext Code on page 102
Total Length
The total number of sent digits. Accepts up to 16 digits.
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Chapter 23: CDR system parameters
Administers the Call Detail Recording (CDR) feature used to record information on incoming, outgoing, and tandem calls for each trunk group administered for CDR, including auxiliary trunks. The system records information on each trunk-group call and each station-to-station call. Example command: change system-parameters cdr
CDR system parameters: page 1
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CDR system parameters: page 1
Calls to Hunt Group -- Record
Valid Entry member-ext group-ext
Usage Records the extension of the telephone or data terminal where the call terminated. Records the extension that was dialed.
CDR Account Code Length
Valid Entry 1 to 15
Usage
The number of digits to record when a user enters an account code. For some record formats, a long account code overwrites spaces on the record that are usually assigned to other fields.
CDR Date Format
Valid Entry month/day day/month
Usage The format for the date stamp that begins each new day of call records.
CDR retention (days)
Valid Entry 1 to 20
Usage Specify the number of days for which you want to retain the CDR logs. The default value is 20.
To retain the CDR logs, you must:
· Enable the Enable CDR Storage on Disk option.
· Enter DISK in the Primary Output Endpoint field.
Important:
- If you enter CDR1 or CDR2 in the Primary Output Endpoint field, then the CDR logs are not retained on the Primary Controller (main server).
- In case of survivable server, if the survivable server is active, then the CDR logs are retained on the local disk.
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CDR system parameters
Condition Code `T' for Redirected Calls
Enables or disables identification of CDR records of calls that have been redirected automatically off the server running Communication Manager.
Valid Entry y n
Usage The Condition Code of both CDR records for the call is `T.' Special identification is applied.
Digits to Record for Outgoing Calls
Valid Entry dialed outpulsed
Usage
Record the digits a user dials.
Record the digits that Communication Manager sends out over the trunk, including any additions or deletions that take place during routing.
Disconnect Information in Place of FRL
Valid Entry y
n
Usage
Replace the Facility Restriction Level (FRL) with information about why a call disconnects.
Record the call FRL.
Enable CDR Storage on Disk
Enables or disables the Survivable CDR feature for the main server, Survivable Remote Server (Local Survivable Processor), and Survivable Core Server (Enterprise Survivable Server). Default is disabled.
Force Entry of Acct Code for Calls Marked on Toll Analysis Form
Specifies whether or not an account code is required when making a toll call. Account codes are not required on all charged calls, and are required on some non-charged calls.
Valid Entry y
Usage
Restricts all toll calls unless the user dials an account code. Available only if Forced Entry of Account Codes is enabled for the system.
Table continues...
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CDR system parameters: page 1
Valid Entry n
Usage
Users can place a call without an account code. This does not override other calling restrictions.
Related links Forced Entry of Account Codes on page 834
Inc Attd Call Record
Enables or disables separate recording of attendant portions of outgoing calls that are transferred or conferenced.
Available only if Incoming Trunk Call Splitting is enabled.
Related links Inc Trk Call Splitting on page 107
Inc Trk Call Splitting
Enables or disables the creation of separate records for each portion of incoming calls that are transferred or conferenced.
Available only if Record Outgoing Calls Only is disabled.
Related links Record Outgoing Calls Only on page 111
Interworking Feat-flag
Valid Entry y n
Usage The feature flag indicates interworked outgoing ISDN calls. The feature flag indicates no answer supervision for interworked calls.
Intra-Switch CDR
Enables or disables the recording of calls within Communication Manager. Requires administration of intra-switch CDR extensions.
Related links Intra-Switch CDR on page 432
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CDR system parameters
Modified Circuit ID Display
Affects the "printer", "teleseer", and "59-character" output formats.
Valid Entry y
n
Usage Displays the circuit ID in its actual format (100's, 10's, units). For example, circuit ID 123 displays as 123.
Note:
Requires verification that an output device can accept this format. Displays the circuit ID in its default format (10's, units, 100's). For example, the system displays circuit ID 123 as 231.
Node Number (Local PBX ID)
Displays the DCS switch node number in a network of switches.
Outg Attd Call Record
Enables or disables separate recording of attendant portions of outgoing calls that are transferred or conferenced.
Available only if Outgoing Trunk Call Splitting is enabled.
Related links Inc Trk Call Splitting on page 107 Outg Trk Call Splitting on page 108
Outg Trk Call Splitting
Enables or disables the creation of separate records for each portion of outgoing calls that are transferred or conferenced
Primary Output Endpoint
Primary Output Endpoint determines where the server running Communication Manager sends the CDR records. You require Primary Output Endpoint if a Primary Output Format is specified.
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CDR system parameters: page 1
Valid Entry eia Extension number
CDR1, CDR2
DISK
Usage
The EIA port is used to connect the CDR device.
The extension of the data module that links the primary output device to the server running Communication Manager.
The CDR device is connected over a TCP/IP link, and this link is defined as either CDR1 or CDR2 for IP Services.
If you configure the CDR Retention (Days) field, then you must enter DISK in the Primary Output Endpoint field so that the CDR logs are retained.
If you configure the retention policy, the retention policy is applicable if the Primary Output Endpoint field is set to DISK only.
Related links IP Services on page 498
Primary Output Format
Indicates the format of the call records sent to the primary output device.
Valid Entry customized
printer
59-char expanded lsu lsu-expand int-direct int-isdn int-process teleseer unformatted
Usage
For special call accounting needs that are prevented for standard record formats. Requires call accounting software that is also customized to receive these records. Call accounting vendors should be consulted before using this option.
Call detail records are sent to a printer rather than to a record collection or call accounting system.
For standard record formats. The selection must be compatible with your call accounting software. Verify this through your vendor or the accounting system documentation.
Privacy -- Digits to Hide
Indicates how much of the dialed number to hide on the CDR record for stations administered for CDR privacy.
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CDR system parameters
Valid Entry 0 to 7
Usage The number of digits to hide, counting from the end right to left.
Example For a value of 4, when the user dials 555-1234, the system displays only "555" in the CDR record.
Related links CDR Privacy on page 768
Record Agent ID on Incoming
Determines whether the EAS agent login ID mut be included instead of the physical extension in the CDR record.
Available only if Expert Agent Selection (EAS) is enabled for the system. Cannot be enabled if Record Called Vector Directory Number Instead of Group or Member is enabled.
Related links Record Called Vector Directory Number Instead of Group or Member on page 110
Record Agent ID on Outgoing
Determines whether the EAS agent's LoginID must be included instead of the physical extension in the CDR record.
Available only if Expert Agent Selection (EAS) is enabled for the system.
Related links Expert Agent Selection (EAS) on page 843
Record Call-Assoc TSC
Enables or disables the creation of records for call-associated temporary signaling connections. A large number of data connections could increase the number of records, so call collection device capacity must be taken into consideration.
Record Called Vector Directory Number Instead of Group or Member
Determines whether or not to include the Vector Directory Number (VDN) in the CDR record. If enabled, the called VDN overrides the group or member information that normally the system displays in the CDR record. If a call is directed through more than one VDN, the first VDN used for
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CDR system parameters: page 1
the call is stored. This applies only to calls routed to a hunt group by a vector, not to calls routed directly to an extension by a vector. Cannot be enabled if Record Agent ID on Incoming is enabled. Related links Record Agent ID on Incoming on page 110
Record Non-Call-Assoc TSC
Enables or disables the creation of records for non-call-associated temporary signaling connections. A large number of data connections could increase the number of records, so record collection device capacity must be taken into consideration.
Record Outgoing Calls Only
Valid Entry y
n
Usage Record only outgoing calls.
Note: This can save space if you are only concerned with charges for outbound calls. Record both outgoing and incoming calls.
Remove '+' from SIP Numbers
If you set the Remove '+' from SIP Numbers field on the CDR System Parameters screen to y and if a leading plus (+) sign is present in a calling or called number sent over a CDR adjunct, Communication Manager removes the leading plus (+) sign from the number. The default value is y.
Remove # From Called Number
Valid Entry y
n
Usage
Removes the "#" or "E" symbol from the Dialed Number field of the call detail record. The output device must be able to accept this format.
Retains the trailing "#" or "E" symbol in the Dialed Number field whenever an interdigit time-out occurs or users dial # to indicate the end of dialing.
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CDR system parameters
Secondary Output Endpoint
Available only if Secondary Output Format is administered.
Valid Entry eia Extension number
CDR1, CDR2
Usage
The secondary output device is connected to the eia port.
The extension of the data module that links the secondary output device to the server running Communication Manager.
The CDR device is connected over a TCP/IP link, and this link is defined as either CDR1 or CDR2 for IP Services.
Related links Secondary Output Format on page 112 Service Type on page 500
Secondary Output Format
Caution:
Only qualified Avaya service personnel should administer a secondary output device. This option might cause loss of data when the buffer contains large amounts of data.
Valid Entry customized int-direct int-process lsu unformatted
Usage
The format of the call records sent to the secondary output device. Only these formats can be used for a secondary output device. The format must be compatible with call accounting software. Verify this through the vendor or the accounting system documentation.
Suppress CDR for Ineffective Call Attempts
Ineffective call attempts are the calls that are blocked because the user has insufficient calling privileges or all outgoing trunks are busy. These calls appear on the CDR record with a condition code "E".
Valid Entry y
Usage
Ignores ineffective call attempts. Used when there is limited storage space for CDR records and when records often overrun the buffer.
Table continues...
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Valid Entry n
CDR system parameters: page 1
Usage Reports ineffective call attempts. Used when users are often unable to place outgoing calls, or if a large number of incoming calls are incomplete. Also used to keep records of attempts to contact a client, and when ISDN trunks are used. This option requires more space for records.
Use Enhanced Formats
Enables or disables the use of the Enhanced version of the specified primary output formats. Enhanced formats provide additional information about time in queue and ISDN call charges, where available. This affects the expanded, teleseer, lsu, printer, and unformatted output formats. Enhanced formats and ISDN formats cannot be used at the same time.
Use ISDN Layouts
Enables or disables using the ISDN version of the specified primary output format. ISDN Layouts provide more accurate information about the inter-exchange carrier and ISDN network services used for a call. This affects lsu and printer output formats, as well as any format with ISDN layouts, such as teleseer. ISDN formats and Enhanced formats cannot be used at the same time.
Use Legacy CDR Formats
Enables or disables the use of pre-Communication Manager 4.0 legacy Call Detail Recording (CDR) formats for CDR records. The default is enabled. Listed below are the CDR formats that are impacted by this field. All other CDR formats remain unchanged.
CDR Format
ISDN Teleseer Enhanced Teleseer ISDN Printer Enhanced Printer ISDN LSU Enhanced LSU Expanded Enhanced Expanded Unformatted Enhanced Unformatted Int-ISDN
Communication Manager 3.1 and earlier length 80 81 84 85 59 59 135 151 105 119 136
Communication Manager 4.0 and later length 82 83 86 87 61 61 139 155 109 123 140
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CDR system parameters
CDR System Parameters: page 2
Used if there is an arrangement with the vendor to customize the call accounting system to receive these records. Available only if Primary Output Format is customized. Related links Primary Output Format on page 109
Data Item
Use this field to itemize the data items that appear on the customized record.
You must include at least one field for a record. The last two data items in the record must be linefeed and return.
Data Item acct-code attd-console auth-code bandwidth bcc calltype calling-num cca-id clg-pty-cat clg-num/in-tac code-dial code-used cond-code contact-uri country-from country-to date dialed-num duration end-date(4d) feat-flag
Length 15 2 7 2 1 1 15 20 2 10 4 4 1 20 3 3 6 23 4 8 1
Table continues...
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Data Item frl from-uri in-crt-id in-trk-code ins internal-codec isdn-cc isdn-cc/ppm ixc-code line-feed local-time-from local-time-to location-from location-to ma-uui node-num null out-crt-id ppm request-uri res_flag return sec-dur seq-num space start-date start-date(4d) start-time time timezone-from timezone-to to-uri tsc_ct tsc_flag trunk-codec
Length 1 20 3 4 3 2 11 11 4 1
3 3 1 2 3 3 5 20 1 1 5 10 1 6 8 6 4 3 6 20 4 1 2
CDR System Parameters: page 2 Table continues...
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CDR system parameters
Data Item ucid vdn
Length 20 5
Length
The length of each data item. The maximum record length depends on the call accounting system. Check with the vendor. The date field should be six digits to ensure proper output. Certain fields default to the required length.
Record Length
Displays the accumulated total length of the customized record, updated each time the length of a data item changes.
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Chapter 24: Change Station Extension
Changes extensions on the switch from one extension to another all at once. Specifically: · All administration that was associated with the current extension is associated with the new extension. · Any administration references on the changing extension, such as references used in a vector, coverage, and so on, are now referenced to the new extension. · All references to the previous extension are removed from the system.
If an extension is changed that is also administered on an adjunct, the extension on the adjunct must also be changed to ensure proper functionality. For fields that are changing, current information prior to the change displays under From Extension. A forwarded extension administered as a button remains. Exceptions Station extensions cannot be changed if that station is administered as the emergency location extension for another station. For example, if station A is administered as the emergency location extension for station B, then:
· First assign station B to a different emergency location extension. · First change station B. The system displays station A's extension as the emergency location
extension. Example command: change extension-station n, where n is the extension assigned to a station.
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Change Station Extension
Emergency Location Extension
The Station screen displays the new extension for the Emergency Location Ext. field. Up to sixteen numbers can be used to make up a valid extension number for the dial plan.
IP Parameter Emergency Location
The emergency location extension from the IP Address Mapping screen associated with the current extension. Related links IP network region on page 459 Emergency Location Extension on page 433
Message Lamp
The new extension for the message lamp extension. Up to sixteen numbers can make up a valid extension number for the dial plan.
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Station
Station
The new extension. Up to seven numbers can make up a valid extension number for the dial plan.
Station Name
The name of the existing extension.
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Chapter 25: Circuit Packs
For more information, see Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and Servers.
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Chapter 26: Class of Restriction
Establishes classes of restriction (COR). Classes of restriction restrict users from originating or terminating certain types of calls. A system might use only one COR or as many as necessary to control calling privileges. Example command: change cor n, where n is the COR number.
Class of restriction: page 1
Access to MCT
Grants or denies access to the Malicious Call Trace feature.
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Class of Restriction
Valid Entry y n
Usage
With this, user can activate a request to trace a malicious call.
With this, user cannot request a malicious call trace. If this extension is the subject of a malicious call trace, the MCT History report does not display the extension.
Add/Remove Agent Skills
User can add or remove skills.
APLT
Users can gain access to APLT trunk group Enhanced Private Switched Communications System (EPSCS) or Common Control Switched Arrangement (CCSA) off-net facilities.
Automatic Charge Display
Shows the cost of an active outgoing call using Periodic Pulse Metering (PPM) or ISDN Advice of Charge (AOC) on Digital Communications Protocol (DCP) or Avaya BRI stations. Unavailable in the U.S.
Valid Entry y n
Usage
Call charges display during and at the end of the call.
Call charges display only when users press the disp-chrg button before the call drops.
Called Party Restriction
Valid Entry Inward
Manual
Public
Termination none
Usage
Blocks the calling party from receiving incoming exchange network calls, attendant originated calls, and attendant completed calls.
Blocks the called party from receiving all calls except for those originated or extended by the attendant.
Blocks the called party from receiving public network calls. Attendant calls can go through the called party and attendant-assisted calls if administered for this COR.
Blocks the called party from receiving any calls at any time.
No called party restrictions.
Related links Restriction Override on page 128
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Class of restriction: page 1
Calling Party Restriction
Determines the level of calling restriction associated with this COR. Important: Limit calling permissions as much as possible.
Valid Entry Origination Outward All-toll
Tac-toll
none
Usage
Blocks the calling party from originating a call from the facility at any time. The party can only receive calls. A telephone with this COR can initiate Remote Access calls, if the COR of the barrier code supports it.
Blocks the calling party from calling outside the private network. Users can dial other users on the same server running Communication Manager or within a private network. To enhance security, you can use outward restrictions when practical.
Blocks the calling party from making ARS and trunk access calls to certain toll areas as defined by the Dialed String on the ARS Toll Analysis table. The call completes if the facility COR is also associated with an Unrestricted Call List and whose Dialed String also matches the dialed number.
Blocks the calling party from making trunk access calls to certain toll areas as defined by the Dialed String on the ARS Toll Analysis table. The call completes if the facility COR is also associated with an Unrestricted Call List and whose Dialed String also matches the dialed number.
No calling party restrictions.
Related links Dialed String on page 21
Can Be a Service Observer
A user with this CO permission can be a service observer.
Available only if Basic Service Observing is enabled for the system.
Caution:
Service Observing might be subject to federal, state, or local laws, rules, or regulations; or require the consent of one or both of the parties to the conversation. Customers should familiarize themselves with and comply with all applicable laws, rules, and regulations before using these features.
Related links Service Observing (Basic) on page 845
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Class of Restriction
Can Be Picked Up By Directed Call Pickup
Calls for this station or EAS agent can be picked up using the Directed Call Pickup Up feature. Available only if Directed Call Pickup is enabled for the system. Related links Directed Call Pickup on page 363
Can Be Service Observed
Service observing can be activated for physical extensions and logical agent IDs and VDNs.
Can Change Coverage
Station users can: Or Station users can have permission to: · Select one of two previously-administered coverage paths · Activate, change, or deactivate call forward all calls or call forward busy/don't answer from any on-site or off-site location
Can Use Directed Call Pickup
The station, attendant, or EAS agent permission can pick up calls using the Directed Call Pickup feature. Available only if Directed Call Pickup is enabled for the system. Related links Directed Call Pickup on page 363
COR Description
The description of the COR. Accepts up to 35 characters. Tip: A clear description makes it easier to remember which COR to assign when adding users.
Example Two examples of COR descriptions include "Customer Service" and "Legal Department".
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Class of restriction: page 1
COR Number
The COR number.
Direct Agent Calling
Users can use the Direct Agent Calling feature. Using Direct Agent Calling, users can dial an ACD agent extension directly, rather than anyone in the agent pool. If the system is in Night Service, the call routes to the Night Service extension. If the extension with this COR belongs to an agent, the agent can receive calls directly.
Facility Access Trunk Test
Users can perform Facility Access Trunk Tests. When this feature is active, the trk-ac-alm feature button status lamp lights when a successful test attempt occurs.
Forced Entry of Account Codes
If enabled, the Forced Entry of Account Codes (FEAC) feature is active on this COR. Any telephone assigned the associated COR must dial an account code before making an outgoing call. If a call is being routed by ARS, the system prevents account code checking on the COR. Available only when the FEAC feature is enabled for the system.
Important: If a COR using the FEAC feature is assigned to a VDN, the route-to commands executed by the associated vector will be unsuccessful. Related links Forced Entry of Account Codes on page 834 CDR FEAC on page 892
FRL
Security alert: Assign the lowest possible FRL to enhance system security.
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Class of Restriction
Valid Entry 0 to 7
Usage
Assigns a Facilities Restriction Level (FRL) to the COR to enhance system security. AAR or ARS features use this entry to determine call access to an outgoing trunk group. Outgoing call routing is determined by a comparison of the FRLs in the AAR/ARS Routing Pattern and the FRL associated with the COR of the call originator. The call originator is typically a telephone user. An originating FRL of 0 has the least calling privileges.
Fully Restricted Service
Unavailable for Enhanced Private Switched Communications System (EPSCS) or Common Control Switched Arrangement (CCSA) off-net facilities.
Valid Entry y
n
Usage
Stations do not have access to the public network for either incoming or outgoing calls.
Stations have access to the public network for either incoming or outgoing calls.
Group II Category For MFC
Valid Entry 1 to 10
Usage
Controls categories for Russian signaling trunks. Also controls categories for R2MFC signaling trunks if administered system-wide. The server running Communication Manager sends this value as the Calling or Called Party Category for telephones or trunks that use this COR. This administered Calling Party Category digit is included as part of the ANI information sent to the local telephone company central office on request using R2-MFC signaling.
Related links Use COR for Calling Party Category on page 608
Group Controlled Restriction
Valid Entry active inactive
Usage The current COR is under controlled restriction. The current COR is beyond controlled restriction.
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Class of restriction: page 1
Hear System Music on Hold
If you enable the Music on Hold feature on the telephone, callers that the user puts on hold will hear music.
Hear VDN of Origin Announcement
Agents can receive VDN of Origin Announcement (VOA) messages that provide agents with a short message about the caller's city of origin or requested service based on the VDN used to process the call.
MF ANI Prefix
The prefix applied to an extension number when ANI is sent to the local telephone company central office. Accepts up to seven digits. This COR-specific value overrides any ANI prefix administered system-wide for multifrequency signaling. The prefix is inapplicable when ANI is tandemed through the Communication Manager server on tandem calls. Also applies to the ANI for the server when the originating side is a trunk and there was no ANI.
Related links ANI Prefix on page 601
Partitioned Group Number
Available only if AAR/ARS Partitioning is enabled for the system and partition groups are administered.
Valid Entry 1 to 8
Usage The AAR/ARS partitioned group number associated with this COR.
Related links ARS/AAR Partitioning on page 829
PASTE (Display PBX Data on telephone)
Use this field to enable the PASTE feature.
Valid entry y n
Usage Downloads all lists. Disables the PASTE feature.
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Class of Restriction
Priority Queuing
If enabled, a telephone user's calls are placed ahead of non-priority calls in a hunt group queue. Related links
ACD on page 842
Restricted Call List
Use this field to allow or restrict access to the Restricted call list.
Valid entry y
n
Usage
Users cannot make calls to numbers on the Restricted call list from a facility assigned to this COR.
Users can make calls to numbers on the Restricted call list from a facility assigned to this COR.
Related links RCL on page 894
Restriction Override
Users assigned to this COR can override inward restriction during conference, transfer or call forwarding operation to a telephone that is inward restricted.
Valid Entry attendant
all none
Usage
Attendants might override inward restrictions. A telephone with a COR that is inward restricted cannot receive public network, attendant-originated, or attendant-extended calls.
All users can override inward restrictions
No users can override inward restrictions
Send ANI for MFE
Applicable only: · For Spain · For 2/6 signaling, but not 2/5 signaling
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Valid Entry y
n
Class of restriction: page 1
Usage Enables Automatic Number Identification (ANI) to send the calling party's number to the public or IBERCOM network so that charges are broken down by line. Charges are not itemized by line. The company receives a single bill for the total number of calls made; also called block charging.
Time of Day Chart
Available only if Time of Day Routing is enabled for the system.
Valid Entry 1 to 8
Usage The AAR/ARS time-of-day-chart number associated with this COR.
Related links Time of Day Routing on page 841
Unrestricted Call List
Valid Entry 1 to 10 blank
Usage
Overrides specified toll call restrictions for a COR otherwise restricted from making ARS or trunk access calls.
Related links Dialed String on page 21
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Class of Restriction
Class of Restriction: page 2
ASAI Uses Station Lock
Valid Entry y
n
Usage
Enables the ASAI originated calls to follow the restriction or permission assigned to the locked station, via the Station Lock COR feature.
Enables the ASAI originated calls to follow the permission or restrictions of the COR assigned to the endpoint. The default value is n.
Block Enhanced Conference/Transfer Display
Valid entry y
n
Usage
Blocks all the enhanced conference and transfer display messages on digital telephones, except the Transfer Completed message.
Does not block the enhanced conference and transfer display messages.
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Class of Restriction: page 2
Block Transfer Displays
Valid Entry y
n
Usage
Users of DCP, Hybrid, ISDN-BRI, or wireless display telephones cannot receive a confirmation message when they transfer a call.
Users can receive a confirmation message when they transfer a call.
Brazil Collect Call Blocking
If enabled, all Brazilian trunk calls that terminate to a station send back a double answer to the local telephone company central office (CO). This double answer tells the CO that this particular station cannot accept collect calls. The CO then tears down the call if it is a collect call.
Line Load Control
Valid Entry 1 to 4
Usage
Specifies the line load control level for this COR, where 1 has no restrictions and 4 is the most restrictive.
Maximum Precedence Level
Assigns a maximum precedence level for extensions with this COR for use with the Multiple Level Precedence and Preemption feature.
Valid Entry fo fl im pr ro
Usage Flash Override Flash Immediate Priority Routine. This is the default value.
MF Incoming Call Trace
If enabled, a Call Trace can be assigned to a station. Communication Manager then generates an MFC backward signal during call setup instead of the "free" signal. This triggers the local telephone company central office to collect trace information before releasing the calling party.
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Class of Restriction
MLPP Service Domain
Valid Entry 0 to 16777215
Usage The service domain for users and trunks to which this COR is assigned
This field is visible only if Multi level precedence preemption is set to y on the system-features customer-options form.
Outgoing Trunk Alerting Timer (minutes)
Applies an alerting tone to an outgoing trunk call after an administrable amount of time.
Valid Entry 2 to 999 blank
Usage
The number of minutes to wait before an alerting tone is applied to the call.
The Alerting Tone for Outgoing Trunk Calls feature is disabled and the alerting tone is not applied to the call. This is the default.
Related links Trunk Alerting Tone Interval (seconds) on page 310
Outgoing Trunk Disconnect Timer (minutes)
Disconnects an outgoing trunk automatically after an administrable amount of time.
Valid Entry 2 to 999
blank
Usage
The number of minutes to wait before automatically disconnecting the call. A warning tone is given to all parties on the trunk call 1 minute before the administered value and a second warning tone is heard 30 seconds later. The call is automatically disconnected 30 seconds after the second warning tone.
Outgoing trunk calls disconnect only when dropped by one or all parties. This is the default.
Preemptable
If enabled, makes extensions with this COR preemptable for Multiple Level Precedence and Preemption calls.
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Class of Restriction: page 2
Remote Logout of Agent
If enabled, users can use a feature access code to logout an idle ACD or EAS agent without being at the agent's telephone.
Restrict Second Call Consult
Use this field to enable the Restrict Second Agent Consult feature. With this feature, Communication Manager allows the agents of Avaya Aura® Contact Center to use only one consult operation at a time. The consult operations include transfer and conference.
Note: If you set the Restrict Second Call Consult field to n, after you dial the Restrict Second Consult feature access code, the feature access code overrides the setting of the Restrict Second Call Consult field on the COR group of the station.
Service Observing by Recording Device
If enabled, the service observer associated with this COR can use an audio recording device, such as the Witness product.
Station-Button Display of UUI IE Data
If enabled, a station user can push a uui-info station button and see up to 32 bytes of ASAIrelated User-User-Information Information Element (UUI-IE) data. Pressing the uui-info button displaces the incoming call/collected digits display. Pressing callr-info re-displays the collected digits. Available only with Call Center release 3.0 or later. Related links Call Center Release on page 843
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Class of Restriction
Station Lock COR
Valid Entry 0 to 995
Usage
If the user dials the station lock Feature Access Code (FAC), the user's extension applies the Station Lock COR for call restrictions instead of the station's usual COR. For information on call restrictions, see the Station Lock section of Avaya Aura® Communication Manager Feature Description and Implementation. This field defaults to the current COR.
Related links Station Lock Activation/Deactivation on page 269
Class of Restriction: page 3
The Send All Calls and Call Forwarding (SAC/CF) Override feature overrides active rerouting. This feature overrides these active rerouting settings:
· Send All Calls (SAC) · Call Forwarding (CF) all · Enhanced Call Forwarding (ECF) unconditional
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Class of Restriction: page 3
Allow Trunk Seizure On Busy Indicator
This field specifies whether the trunk is seized when the user presses the Busy indicator button.
Valid entry y
n
Usage When the user presses the Busy indicator button for TAC: · The trunk is seized. · On pressing the digits, the call goes through. When the user presses the Busy indicator button for TAC: · The trunk is not seized. · The user hears reorder tone with fluttering lamp. · The system logs a denial event.
Auto Answer
Enables or disables automatic answer on team button calls.
Erase 24xx/14xx/94xx User Data Upon: Dissociate or unmerge this phone
Administers what local terminal data items are erased when the 24xx, 14xx, or 94xx is dissociated or unmerged.
Valid Entry none log customizations all
Usage Default value. No local terminal data is erased. Terminal's local call Log data is erased. Call Log, Button labels, Speed Dial List, Local Terminal Options are erased. All local terminal data is erased (Call Log, Button Labels, Speed Dial List, Options, Language).
Erase 24xx/14xx/94xx User Data Upon: EMU login or logoff at this phone
Administers what local terminal data items are erased upon Enterprise Mobility User (EMU) login or logoff.
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Class of Restriction
Valid Entry none log customizations all
Usage This is the default. No local terminal data is erased. Terminal's local call Log data is erased. Call Log, Button labels, Speed Dial List, Local Terminal Options are erased. All local terminal data is erased (Call Log, Button Labels, Speed Dial List, Options, Language).
Mask CPN/Name for Internal Calls
Valid entry y
n
Usage
Hides the display of calling and called party numbers and administered name on internal calls.
Shows the display of calling and called party numbers and administered name on internal calls.
one-X Server Access
Enables or disables the one-X Server Access for one-X control.
Outgoing Trunk Disconnect Timer (minutes)
Disconnects an outgoing trunk automatically after an administrable amount of time.
Valid Entry 2 to 999
blank
Usage
The number of minutes to wait before automatically disconnecting the call. A warning tone is given to all parties on the trunk call 1 minute before the administered value and a second warning tone is heard 30 seconds later. The call is automatically disconnected 30 seconds after the second warning tone.
Outgoing trunk calls disconnect only when dropped by one or all parties. This is the default.
Priority Ring
Enables or disables priority ringing for speed dialing on team button calls.
SAC/CF Override by Priority Call and Dialing
Use this field to enable or disable the SAC/CF override by Priority Call and Dialing feature.
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Class of Restriction: page 3
SAC/CF Override by Team Btn
Use this field to allow the user of a monitoring station to override any active rerouting, such as Send All Calls or Call Forwarding, on the monitored station. The monitoring station can override the rerouting only if the following conditions are met:
· The SAC/CF Override by Team Btn field is set to y. · The SAC/CF Override Protection for Team Btn field is set to n. · The monitoring station uses the team button the make the call to the monitored station. Note: Communication Manager Release 6.2 and earlier supported this feature on H.323 and DCP stations. Communication Manager Release 6.3.6 extends this support to SIP stations.
SAC/CF Override Protection for Priority Call and Dialing
Use this field to enable or disable the SAC/CF Override Protection for Priority Call and Dialing feature.
SAC/CF Override Protection for Team Btn
Use this field to restrict the user of a monitoring station from overriding the active rerouting, such as Send All Calls or Call Forwarding, on the monitored station. This field works only when the monitoring station uses the team button the make a call to the monitored station.
Note: Communication Manager Release 6.2 and earlier supported this feature on H.323 and DCP stations. Communication Manager Release 6.3.6 extends this support to SIP stations.
Team Btn Display Name
Enables or disables the display of station name on team button calls.
Team Btn Silent if Active
Enables or disables audible ringing on team button calls.
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Class of Restriction
Team Pick Up by Going Off Hook
Enables or disables pick-up by going off hook on team button calls.
Class of restriction: page 4 to page 23
You can use the screens from page 4 to page 13 to assign calling permissions and the screens from page 14 to page 23 to assign service observing permissions to the COR that you administer.
CALLING PERMISSION
Users with this COR can call users with the CORs from 0 to 995.
Valid entry y
n
Usage
A station user with the COR that you administer can call stations with the other COR. The default value is y.
A station user with the COR that you administer cannot call stations with the other COR.
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SERVICE OBSERVING PERMISSION
Class of restriction: page 4 to page 23
Users with this COR can observe users with the CORs from 0 to 995.
Valid entry y
n
Usage
A station user with the COR that you administer can observe stations with the other COR. The default value is y.
A station user with the COR that you administer cannot observe stations with the other COR.
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Chapter 27: Class of service
Administers access permissions for call processing features that require dial code or feature button access.
Note: Class of Service (COS) is inapplicable to trunk groups except for the Remote Access feature. A COS assignment defines which features and functions a telephone user can access. Up to 16 different COS numbers can be administered (0 to 15). When Tenant Partitioning is enabled for the system, you can administer up to 100 COS groups, each with 16 Classes of Service. The screen lists the default values for each COS or feature combination. For a particular combination, y is for access to the feature and n is for denying access. Caution: Because many hunt groups are set up with COS 1, be careful when you assign restrictions to COS 1. Example command: change cos-group n, where n is the COS group number. Related links Tenant Partitioning on page 840
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Class of service: page 1
Class of service: page 1
Automatic Callback
Enables or disables Automatic Callback. Using Automatic Callback, internal users who place a call to a busy or an unanswered internal telephone can be automatically called back when the called telephone becomes available.
Automatic Exclusion
Available when Automatic Exclusion is enabled for the system and the applicable station has an assigned Exclusion button.
Valid Entry y
n
Usage
Activates Automatic Exclusion when a user goes off hook on a station. Automatically prevents other multi-appearance users from bridging onto a call.
Denies Automatic Exclusion. Manual exclusion is available when the user presses the Exclusion button before dialing or during a call.
Related links Automatic Exclusion by COS on page 353
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Class of service
Call Forwarding All Calls
Used to activate the call forwarding feature for all calls to any extension.
Call Forwarding Busy/DA
If enabled, users can forward calls to any extension when the dialed extension is busy or does not answer.
Client Room
Users can access Check-In, Check-Out, Room Change/Swap, and Maid status functions. In addition, Client Room is required at consoles or telephones that are to receive message-waiting notification. Available only with Hospitality Services and a Property Management System interface.
Console Permissions
Multi-appearance telephone users can control the same features that the attendant controls.
Contact Closure Activation
A user can open and close a contact closure relay. Contact closures control electrical devices remotely. Users dial a Feature Access Code (FAC) on a telephone to activate electrical devices such as electrical door locks.
COS Group
To administer a COS group, Tenant Partitioning must be enabled for the system.
Valid Entry 1 to 100
Usage The COS Group number.
Related links Tenant Partitioning on page 840
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Class of service: page 1
COS Name
The identifying name for this COS group. To administer a COS group, Tenant Partitioning must be enabled for the system. Related links Tenant Partitioning on page 840
Data Privacy
If activated, a user can enter a Feature Access Code (FAC) that protects a data call from being interrupted by any of the system override or ringing features.
Extended Forwarding All
A user can administer call forwarding from a remote location for all calls. Available only if Extended Coverage and Forwarding Administration is enabled for the system. Related links Extended Cvg/Fwd Admin on page 833
Extended Forwarding B/DA
A user can administer call forwarding from a remote location when the dialed extension is busy or does not answer. Available only if Extended Coverage and Forwarding Administration is enabled for the system. Related links Extended Cvg/Fwd Admin on page 833
Off-Hook Alert
Enables or disables the requirement that the system send an emergency call to the attendant if the telephone remains off-hook for a prescribed time. Available only if Basic Hospitality or Emergency Access to Attendant is enabled for the system. Related links Hospitality (Basic) on page 834 Emergency Access to Attendant on page 833
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Class of service
Personal Station Access (PSA)
Users can associate a telephone with their extension using their programmed services using a Feature Access Code (FAC). Deactivate for virtual telephones. Required for a user's home station in order for that user to use the Enterprise Mobility User (EMU) feature at other stations. Available only if Personal Station Access (PSA) is enabled for the system.
Related links Personal Station Access (PSA) on page 838
Priority Calling
If activated, a user can dial a Feature Access Code (FAC) to originate a priority call. Priority calls ring differently and override send all calls.
QSIG Call Offer Originations
If activated, a user can invoke QSIG Call Offer services.
Restrict Call Fwd-Off Net
Valid Entry y
n
Usage A user cannot forward calls to the public network.
Security alert:
For security reasons, this should be enabled for all classes of service except the ones used for special circumstances. A user can forward calls to the public network.
Trk-to-Trk Restriction Override
Use this field to allow or deny user permission to override the system or COR-to-COR calling party restrictions that prohibit the trunk-to-trunk transfer operation for users with this COS.
Security alert:
Use this COS capability with caution. The ability to perform trunk-to-trunk transfers greatly increases the risk of toll fraud.
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Class of service: page 2
Class of service: page 2
Ad hoc Video Conferencing
Enables or disables Ad-hoc Video Conferencing, so that up to six users can participate in a video conference call.
Bridging Exclusion Override
When enabled, allows a station to bridge onto a call that had exclusion activated.
Call Forwarding Enhanced
If activated, users can designate different preferred destinations for forwarding calls that originate from internal and external callers.
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Class of service
DCC Activation/Deactivation
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
Valid Entry y n
Usage Activate the Destination Code Control feature. Deactivate the Destination Code Control feature. The default value is n.
Masking CPN/Name Override
Users can be assigned or denied users permission to override the MCSNIC capability. MCSNIC masks the display of calling party information and replaces it with a hard-coded system-wide text string, Info Restricted.
Match BCA Display to Principal
Valid entry y
n
Usage
The display format of incoming calls on the bridged call appearances of a COS group is <calling name/number>.
The display format of incoming calls on the bridged call appearances of a COS group is <calling name/number> to <principal station>.
MOC Control
The MOC Control field indicates that the station is under Microsoft Office Communicator (MOC) control.
If a station uses the MOC client and also wants to use an office telephone and an off-PBX telephone, such as a mobile phone simultaneously, set the MOC Control field to y.
Valid entry y n
Usage Station is under MOC control. Station is beyond MOC control. This is the default value.
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Class of service: page 2
Priority Ip Video
Activate or deactivate priority video calling. Video calls have an increased likelihood of receiving bandwidth and can also be allocated a larger maximum bandwidth per call.
VIP Caller
Enables or disables automatic priority calling when assigned to the originator of a call. A call from a VIP telephone is always a priority call without the use of a feature button or FAC.
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Chapter 28: Cluster Session Manager
Use this screen to administer Session Managers. You can administer up to 10 clusters of Session Managers by using the Communication Manager CLI. The node names administered on this screen can be either of IPv4 or IPv6 address. For example, if the primary Session Manager points to an IPv4 type address, then the secondary Session Manager must be IPv4 address type. The node names cannot be a mix of IPv4 and IPv6 address types on the same cluster. To administer cluster Session Manager screen, use the following command:
change cluster session-manager
Cluster Number By default, there are 10 clusters.
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Cluster Name Use this field to add Session Manager node names.
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Chapter 29: Code Calling IDs
On systems with chime paging, assigns a unique series of chimes to extensions using a chime code. The chime code assigned to an extension plays over the speakers whenever that extension is paged. Assigns chime codes to up to 125 extensions. Example command: change paging code-calling-ids
Ext
Assigns extensions to chime codes. Only one extension can be assigned to each chime code. The extension cannot be assigned to a code that is a Vector Directory Number (VDN).
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Chapter 30: Configuration Set
Use the Configuration Set screen to define call treatment options for Extension to Cellular calls. With the Extension to Cellular feature, you can use up to 99 configuration sets that are already defined in the system with default values.
Note: In the One-X mobile environment, you can edit the values of only the Cellular voice mail detection field and the Call log notify field. All other fields are read-only. To modify the values of the fields on the Configuration Set screen, type the following command: change off-pbx-telephone configuration-set n, where n is the Configuration Set number.
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Configuration Set
Apply Ringback upon Receipt of
Use the Apply ringback Upon Receipt of field to configure when Communication Manager must send the ringback tone to the caller.
Valid entry Call-Proceeding
Alert
Usage
Communication Manager sends the ringback tone to the caller after receiving the Call-Proceeding message from the EC500 network. Call-Proceeding is the default value.
Communication Manager sends the ringback tone to the caller after receiving the Alert message from the EC500 network.
Barge-In Tone
Enables or disables a barge-in tone used to add security to Extension to Cellular calls. If a user is on an active Extension to Cellular call and another person joins the call from an Extension to Cellular enabled office telephone, all parties on the call hear the barge-in tone.
Call Appearance Selection for Origination
Specifies how the system selects a Call Appearance for call origination. To use this feature, bridged calls must be enabled for the system.
Valid Entry first-available primary-first
Usage The system searches for the first available regular or bridged Call Appearance. The system first searches for a regular Call Appearance for call origination.
· If all Call Appearances are busy at the primary station, the call cannot be initiated.
· If the primary station is logged-off, a second search is made that includes both regular and bridged Call Appearances.
This is the default.
Related links Bridged Calls on page 793
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Call Log Notify
Call Log Notify
Use this field to update the Avaya one-X® Client Enablement Services logs with actual outpulsed digits and not the digits that the user dials.
Valid entry dialed-digits outpulsed-digits
Usage
When you enter dialed-digits, the endpoint displays the dialed digits.
When you enter outpulsed-digits, Communication Manager analyzes the dialed digits and sends the digits to Client Enablement Services for call logging.
Calling Number Style
Determines the format of the caller ID for calls from a local Communication Manager extension to an Extension to Cellular telephone.
Valid Entry network
pbx
Usage
Provides a display of only 10-digit numbers. For internal calls, the ISDN numbering tables are used to create the calling number and DCS calls use the ISDN calling number if provided. The externally provided calling number is used when available for externally originated calls.
Provides a display of less than 10-digits. Extensions sent as the calling number for all internally- and DCS network-originated calls.
Calling Number Verification
Use this field to impose restrictions on calls originating from an Extension to Cellular-enabled cell phone to some other extension.
Valid entry y
n
Usage
If you set the field to y and the network service provider has not marked the screening indicator as network provided or user provided verified and passed, Communication Manager treats the incoming call as an ordinary incoming trunk call and not as a call from an extension.
If you set the field to n, Communication Manager treats the incoming call as a call from the corresponding desk phone.
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Configuration Set
CDR for Calls to EC500 Destination
Determines whether a Call Detail Record (CDR) is generated for any call to the cell telephone. Available only if CDR reports is enabled for the trunk group.
Valid Entry y n
Usage Treats calls to the XMOBILE station as trunk calls and generates a CDR. Treats calls to the XMOBILE station as internal calls, without generating a CDR.
Related links CDR Reports on page 506
CDR for Origination
Determines the CDR report format when CDR records are generated for a call that originates from an Extension to Cellular cell phone. To generate this CDR, you must enable the Incoming Trunk CDR. The CDR report excludes dialed Feature Name Extensions (FNEs).
Valid Entry phone-number
extension
none
Usage
The calling party on the CDR report is the 10-digit cell phone number. This is the default.
The calling party on the CDR report is the internal office telephone extension associated with the Extension to Cellular cell phone
The system does not generate an originating CDR report.
Cellular Voice Mail Detection
You can use this field if you do not want an Extension to Cellular call to cover to the cellular voice mail. When the call server detects that the call has covered to the cellular voice mail, the call server brings the call back to the server.
Valid entry None Timed
Usage
The call covers to the cellular voice mail. This entry is the default value.
The system displays the Seconds field. You can enter any value from 1 to 9. If the user answers the call on the cellular phone within the configured number of seconds, Communication Manager disconnects the cellular leg of the call. The call continues to ring at the desk phone.
You can use this configuration for GSM, CDMA, and ISDN networks.
Table continues...
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Valid entry Message
Configuration Set Description
Usage The message option works with carriers who use non-ISDN voice mail systems. You must not use this option with ISDN-based voice mail systems.
Configuration Set Description
A description of the purpose of the configuration set. Accepts up to 20 alphanumeric characters. Example "Extension to Cellular handsets"
Confirmed Answer
Enables or disables Confirmed Answer on Extension to Cellular calls for this station. If enabled, requires the user to input a digit to confirm receipt of a call sent to a cellular telephone by the Extension to Cellular feature. Upon answering the incoming call on the cellular telephone, the user hears a dial tone. The user must then press any one of the digits on the telephone keypad. Until the system receives a digit, the system treats the call as unanswered. The length of time to wait for the digit can be administered from 5 to 20 seconds, with a default of 10 seconds. The system plays a recall dial-tone to indicate that input is expected. During the response interval, the original call continues to alert at the desk set and any stations bridged to the call. If the user does not enter a digit before the time-out interval expires, the call is pulled back from the cell telephone.
EC500Delay Deactivation State
Use this field to control sharing of the current EC500Delay state to IX Clients.
IX clients use the EC500Delay feature to delay the EC500 termination on the same device on which it has registered the Avaya Aura® application in VoIP mode.
Valid entry y
n
Usage
Enables sharing the current EC500Delay state, if EC500Delay feature is deactivated due to any feature interaction. This is the default value.
Disables sharing the current EC500Delay state.
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Configuration Set
Fast Connect on Origination
Use this field to specify whether Communication Manager must perform additional processing before connecting an H.323 or a SIP call.
Valid entry y
n
Usage
Communication Manager sends a CONNECT or 200 OK message immediately in response to the SETUP or the INVITE message. The call is connected immediately to the destination. Communication Manager does not send the CONNECT or 200 OK message immediately in response to the SETUP or the INVITE message. The call is connected only when the digits in the SETUP or the INVITE message resolve to an extension or a feature access code.
Feature Invocation by In-Call DTMF Code
Use this field to enable EC500 users to use the in-call features, such as Hold and consult, Transfer, and Conference.
Valid entry y n
Usage Enables EC500 in-call feature invocation. Disables EC500 in-call feature invocation.
Location for routing incoming overlap calls
With the Location for routing incoming overlap calls feature, the administrator can configure the location to route incoming calls that are mapped to EC500, CSP, and Avaya one-X applications over ISDN-PRI, ISDN-BRI, and H.323 overlap trunks.
The overlap trunk refers to an ISDN trunk group configured for overlap receiving. The Digit Handling (in/out) field on the ISDN Trunk Group screen is set to overlap/overlap or overlap/enbloc.
Valid entry Station-location-if-set
Usage
Communication Manager uses the station location to process all calls coming from an EC500-mapped phone. Communication Manager uses the station location only if the location is configured on the EC500-mapped station screen. This entry is the default value.
Table continues...
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Valid entry Trunk
Post Connect Dialing Options
Usage Communication Manager uses the incoming trunk location to process all calls coming from an EC500-mapped phone.
Post Connect Dialing Options
Determines whether additional capabilities, beyond standard ISDN dialing, are available for those incoming ISDN trunk calls that are mapped into XMOBILE stations. These options come into effect after the call has entered the active state (Communication Manager has sent a CONNECT message back to the network).
Valid Entry dtmf out-of-band
both
Usage
Expect digits from either in-band or out-of-band, but not simultaneously. The server allocates a DTMF receiver whenever it needs to collect digits. This option normally would be used for Extension to Cellular XMOBILE station calls.
Expect all digits delivered by out-of-band signaling only. Communication Manager collects digits that it needs from the out-of-band channel (no touch-tone receiver). In addition, any digits received when the server is not collecting digits are converted to DTMF and broadcast to all parties on the call. This option is in force for DECT XMOBILE station calls.
Expect all subsequent digits delivered by simultaneous in-band and out-of-band signaling. Out-of-band signaling consists of digits embedded in ISDN INFO messages while the in-band signaling consists of DTMF in the voice path. Communication Manager collects all digits that it needs from the out-of-band channel. No touch tone receive is allocated to prevent collecting double digits. Endto-end signaling occurs transparently to the server via in-band transmission of DTMF. This option is in force for PHS XMOBILE station calls.
Provide Forced Local Ringback for EC500
When a caller calls an enterprise user, the caller expects a standard ringback tone. If the EC500 feature is activated on the telephone of the enterprise user and if the user uses custom ringback tones on the cell phone, the caller hears the custom ringback tone. If you set the Provide Forced Local Ringback for EC500 field to y, Communication Manager provides the local ringback tone and ignores the custom ringback tone of the cell phone.
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Chapter 31: Console parameters
Administers attendant console group parameters. This includes basic parameters for Centralized Attendant Service (CAS) and Inter-PBX Attendant Service (IAS). A list of the administered attendant consoles also displays. Example command: change console-parameters
Console parameters: page 1
AAR/ARS Access Code
An optional AAR/ARS access code used to route to the main switch. Accepts an up to four-digit access code. Available only for a QSIG branch CAS configuration.
Related links CAS on page 159
Allow Night Service Calls To Forward
Enables or disables forwarding of DID calls to station users. When Station A receives a DID call and Station A does not answer the call, DID times out and the system routes the call to an attendant. When the attendant is in Night Service mode, the system routes the call to a night service station. If Call Forward is active on the night service station, the system routes the call to Station B. When Call Forward is inactive, the night service station will continue to ring.
Valid Entry y n
Usage Enables forwarding of DID calls from the night service station. Disables the forwarding of DID calls to station users. This is the default value.
Alternate FRL Station
The extension of the alternate facility restriction level (FRL) activation station.
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Console parameters: page 1
Attendant Group Name
A name for the attendant group. Accepts up to 27 alphanumeric characters.
Attendant Lockout
Activates or deactivates Privacy -- Attendant Lockout. Privacy -- Attendant Lockout prohibits an attendant from reentering a conference call that has been placed on hold unless recalled by a telephone user on the call. This feature provides privacy for parties on a multiple-party call held on the console. The held parties can hold a private conversation without interruption by the attendant.
Attendant Vectoring VDN
Assigns the VDN extension for Attendant Vectoring to a console. Available only if Attendant Vectoring is enabled for the system and Tenant Partitioning is disabled. Related links Attendant Vectoring on page 830 Tenant Partitioning on page 840
Backup Alerting
Used to assign or remove permissions for system users to pick up alerting calls if the attendant queue has reached its warning state.
Calls In Queue Warning
The number of incoming calls that can be in the attendant queue before the console's second Call Waiting lamp lights. The queue maximum depends on system capacities.
CAS
Using the Centralized Attendant Service (CAS), users at separate locations can concentrate attendant positions at one location. Incoming trunk calls to unattended branch locations are routed to the main attendant.
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Console parameters
Valid Entry main branch
none QSIG-main QSIG-branch
Usage
The main Communication Manager server where the attendant group is located. Uses non-ISDN signaling. CAS Main must be enabled for the system.
A branch Communication Manager server. There are no local attendants, so attendant-seeking calls route to the main Communication Manager server. Uses non-ISDN signaling. CAS Branch must be enabled for the system
Disables CAS.
Same as main, but with QSIG signaling among the Communication Manager servers. Centralized Attendant must be enabled for the system.
Same as branch, but with QSIG signaling among the Communication Manager servers. Centralized Attendant must be enabled for the system.
Related links Centralized Attendant on page 848 RLT Trunk Group No. on page 163
CAS Back-Up Ext
Valid Entry
Usage
Extension number
Individual attendant console
Hunt group
Extension in the dial plan that handles attendant-seeking calls if the RLT trunk group to the CAS Main server is out of service or if CAS Back-Up is activated. Neither a prefixed extension nor a VDN extension is supported. Available only for the CAS Branch feature.
TEG
Related links CAS Branch on page 831 CAS Main on page 831 Centralized Attendant on page 848
COR
Valid Entry 0 to 995
Usage
The class of restriction (COR) number for all attendant consoles. The COR for the individual Attendant Console overrides this value.
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Console parameters: page 1
COS
Valid Entry 1 to 15
Usage
The class of service (COS) number that reflects the required features for all your attendant consoles. The COS for the individual Attendant Console overrides this value.
DID-LDN Only to LDN Night Ext
Valid Entry y
n
Usage
Only listed directory number (LDN) calls go to the listed directory night service extension.
All attendant seeking calls route to the LDN night service extension.
Ext Alert Port (TAAS)
The seven-digit port number assigned to the external alerting device. This supports the Night Service -- Trunk Answer From Any Station feature.
An x indicates that there is no hardware associated with this port assignment. If an x is used, there should also be a value for Ext Alert (TAAS) Extension.
Related links Ext Alert (TAAS) Extension on page 161
Ext Alert (TAAS) Extension
This extension is used by the Terminal Translation Feature (TTI) to assign a port to the Ext Alert Port from a station on the Ext Alert port during system installation or provisioning. Once a port is assigned either through TTI or by entering the Ext Alert Port, the extension is automatically removed and treated as unassigned. Available only when the Ext Alert Port (TAAS) has associated hardware.
Related links Ext Alert Port (TAAS) on page 161
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Console parameters
IAS Att. Access Code
The extension number of the attendant group at the main server running Communication Manager. Required when IAS Branch is enabled. Unavailable if the Centralized Attendant feature is enabled.
Related links IAS (Branch) on page 162 Centralized Attendant on page 848
IAS (Branch)
Enables or disables the Inter-PBX Attendant Service (IAS) Branch feature. Unavailable if Centralized Attendant is enabled.
Related links Centralized Attendant on page 848
IAS Tie Trunk Group No.
Unavailable if the Centralized Attendant feature is enabled.
Valid Entry 1 to 2000
Usage
The number of the tie trunk group to the main for the IAS (Branch). Required when IAS Branch is enabled.
Related links Centralized Attendant on page 848
Night Service Act. Ext.
The extension of the current night service activation station, if any. The station is administered by assigning it a night-serv button.
QSIG CAS Number
Contains the complete number of the attendant group at the main server running Communication Manager, or a Vector Directory Number (VDN) local to the branch server. Accepts up to 20 digits. Cannot be left blank. Available only for an QSIG-branch CAS configuration.
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Related links CAS on page 159
Console parameters: page 1
RLT Trunk Group No.
The trunk group number corresponding to the Release Link Trunk (RLT) trunk group to the main location when supporting CAS Branch service. Available only for a branch CAS configuration.
Related links CAS on page 159
SIP
Use this field to configure SIP support on attendant consoles.
Note:
If native attendant like type "302" or "console" is already configured in the same Attendant Group, then the Communication Manager will not allow you to enable SIP on Console Parameters form. Also, if SIP is enabled on the Console Parameters form, Communication Manager will not allow you to add a new attendant user of type "302" or "console" in the same Attendant Group. Before SIP is enabled, Attendant Vectoring VDN field must be blank.
Valid Entry n y
Usage
Disables SIP attendant. This is the default value.
Enabling the SIP Attendant field will cause attendant seeking calls to route to the designated number assigned in the associated field. When you change the value to y, then the following values appear:
· Routing Digits
· Attendant Server
· Return Call timeout on page 2
When you set the SIP value to y, ensure that:
· The Tenant Partitioning feature is disabled on the system parameters customer options screen.
· All native attendants are removed, if available.
· Communication Manager blocks new attendant if SIP is enabled on console parameters screen.
Routing Digits Access code to route calls to SIP attendant server. You can enter up to four digits only.
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Console parameters
The Routing Digits field appears on the Console Parameters form, only if the Tenant Partition is disabled.
Valid Entry digits from 0 to 9
blank
Usage
Digits comprising the AAR or ARS access code. The default value is blank.
Call routing to attendant server is not specified.
Attendant Server Attendant server number. This field cannot be blank. The Attendant Server field appears on the Console Parameters form, only if the Tenant Partition is disabled.
Valid Entry digits from 0 to 9
Usage You can enter up to 17 digits.
Console parameters: page 2
ABBREVIATED DIALING
List1, List2, List3
Assigns up to three abbreviated dialing lists to each attendant. A personal list cannot be assigned to an attendant.
Valid Entry enhanced group
system
Usage
The attendant can access the enhanced system abbreviated dialing list.
The attendant can access the specified group abbreviated dialing list. A group number is required.
The attendant can access the system abbreviated dialing list.
SAC Notification
Enables or disables Enhanced Attendant Notification for Send All Calls.
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Console parameters: page 2
COMMON SHARED EXTENSIONS
Busy Indicator for Call Parked on Analog Station Without Hardware?
Enables or disables the Busy Indicator lamp that lights for incoming calls parked on Administration Without Hardware (AWOH) stations.
Count
Valid Entry 1 to 1182 blank
Usage
The number of consecutive extensions, beginning with the Starting Extension used as common, shared extensions
Example If you enter a starting extension of 4300 and a count of 3, the system provides three consecutive extension numbers (4300, 4301, and 4302) for parking calls.
Related links Starting Extension on page 165
Starting Extension
The first extension number in a group of consecutive extensions that can be used by the attendant to park calls.
Related links Count on page 165
INCOMING CALL REMINDERS
Alerting (sec)
The number of seconds after which a held or unanswered call is disconnected from an attendant loop and routed to another attendant or night service.
No Answer Timeout (sec)
Valid Entry 10 to 1024 blank
Usage
The number of seconds a call to the attendant can remain unanswered without invoking a more insistent sounding tone. With this option, the call rings for 5 seconds at all points in a coverage path to ensure that the entire path is completed before the call returns to the console.
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Console parameters
Secondary Alert on Held Reminder Calls?
Valid Entry y n
Usage
Begin attendant alerting for Held Reminder Calls with secondary alerting.
Have held reminder calls alert the attendant the same as normal calls. Normal calls start with primary alerting and then switch to secondary alerting when an administered timeout expires.
Related links No Answer Timeout (sec) on page 165
TIMING
Overview timer to Group Queue (sec)
Valid Entry 10 to 1024
blank
Usage
The number of seconds a returning call queues to the individual attendant before overflowing to the group. The value applies if the attendant who previously handled the call is busy or unavailable.
The call immediately goes to the group.
Return Call Timeout (sec)
Valid Entry 10 to 1024 blank
Usage
The time in seconds before a split away call (call extended and ringing a station or otherwise split away from the console) returns to the console. With this option, the call rings for 5 seconds at all points in a coverage path to ensure that the entire path is completed before the call returns to the console. This field applies even when SIP is set to `y' on Console Parameters: page 1.
Time In Queue Warning (sec)
Valid Entry 9 to 999 blank
Usage
The number of seconds a call can remain in the attendant queue before activating an alert.
Time Reminder on Hold (sec)
Valid Entry 10 to 1024
Usage The number of seconds a call can remain on Hold.
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Console parameters: page 3
Console parameters: page 3
Call-Type Ordering Within Priority Levels?
Groups calls to the attendant in the following order: 1. Queue priority level 2. Call type 3. Order received
Call type Type 1 call
Type 2 call Type 3 call
Description
Outgoing public-network calls receiving answer supervision when the Answer Supervision Timer of the trunk group expires, even if the trunk is actually still ringing. Also, incoming calls when answered by the attendant.
Incoming external public-network calls before they receive answer supervision or before the Answer Supervision Timer of the trunk group expires.
All other calls (internal calls, conference calls, and tie-trunk calls of any type). Note that external public-network calls have priority over all other calls including conference calls. Answered public-network calls have priority over unanswered calls.
Valid Entry y
n
Usage
Orders calls by call type within each queue priority category. A type-disp button is assigned to the Attendant Console. The attendant can review the call type for the active call.
Calls are queued in chronological order by queue priority level.
QUEUE PRIORITIES
Assigns a priority level from 1 through 13 to each call category when the call cannot be immediately terminated to an attendant. The calling party hears ringback until an attendant answers the call.
The same priority level can be assigned to more than one call.
Assistance Call
Valid Entry 1 to 13
Usage
Assigns a priority level for assistance calls. A call from a telephone user who dials the attendant-group access code, or from a telephone that has the Manual Originating Line Service feature activated.
Priority 1 is the highest priority.
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Console parameters
DID to Attendant
Valid Entry 1 to 13
Usage
Assigns a priority level for DID to Attendant calls that are incoming DID trunk calls to an attendant group. This excludes trunk calls that return to the attendant group after a timeout or deferred attendant recall.
Priority 1 is the highest priority.
Emergency Access
Valid Entry 1 to 13
Usage Assigns a priority level for emergency access calls that are calls from a telephone user who dials the emergency access code. The default is 1.
Priority 1 is the highest priority.
Individual Attendant Access
Valid Entry 1 to 13
Usage
Assigns a priority level for Individual Attendant Access calls that are calls from a telephone user, incoming trunk call, or a system feature to the Individual Attendant Access (IAA) extension of a specific attendant. If the attendant is busy, the call queues until the attendant is available.
Priority 1 is the highest priority.
Interposition
Valid Entry 1 to 13
Usage Assigns a priority level for Interposition calls that are calls from one attendant to the Individual Attendant Access (IAA) extension of another attendant.
Priority 1 is the highest priority.
Miscellaneous Call
Valid Entry 1 to 13
Usage Assigns a priority level for any unlisted calls. Priority 1 is the highest priority.
Redirected Call
Valid Entry 1 to 13
Usage Assigns a priority level for any calls assigned to one attendant, but redirected to the attendant group because the attendant is now busy.
Priority 1 is the highest priority.
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Console parameters: page 3
Redirected DID Call
Valid Entry 1 to 13
Usage
Assigns a priority level for any DID or ACD calls that time out due to ring/no-answer, busy condition (if applicable), or Number Unobtainable. Calls are rerouted to the attendant group.
Priority 1 is the highest priority.
Return Call
Valid Entry 1 to 13
Usage Assigns a priority level for any calls returned to the attendant after timing out. If the attendant is now busy, the call redirects to the attendant group.
Priority 1 is the highest priority.
Serial Call
Valid Entry 1 to 13
Usage
Assigns a priority level for any calls from the Attendant Serial Call feature when an outside trunk call (designated as a serial call by an attendant) is extended to and completed at a telephone, and then the telephone user goes on-hook. If the attendant who extended the call is busy, the call redirects to the attendant group.
Priority 1 is the highest priority.
Tie Call
Valid Entry 1 to 13
Usage
Assigns a priority level for incoming TIE trunk calls (dial-repeating or direct types) to an attendant group. This excludes trunk calls that return to the attendant group after a timeout or deferred attendant recall.
Priority 1 is the highest priority.
VIP Wakeup Reminder Call
Valid Entry 1 to 13
Usage Assigns a priority level for VIP Wakeup Reminder Calls from the Hospitality feature that send a wake-up reminder to the attendant to call a hotel or motel room.
Priority 1 is the highest priority.
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Console parameters
Console parameters: page 4
QUEUE PRIORITIES
Flash
Valid Entry 1 to 17
Usage The queue priority for Flash precedence level calls.
Flash Override
Valid Entry 1 to 17
Usage The queue priority for Flash Override precedence level calls.
Immediate
Valid Entry 1 to 17
Usage The queue priority for Immediate precedence level calls.
Priority
Valid Entry 1 to 17
Usage The queue priority for Priority precedence level calls.
Console parameters: page 5
Note: If MLPP is disabled, the system does not display the MLPP Queues page, but displays page with the following message: Use the 'list attendant' command to see all administered attendants.
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Chapter 32: Coverage Answer Group
Establishes Call Coverage Answer Groups. An answer group contains up to 100 members who act as a coverage point for another user. For example, if several secretaries are responsible for answering a department's redirected calls, all the secretaries could be assigned to an answer group. The answer group is assigned a group number, and the department's coverage path displays that group number. All telephones in an answer group ring (alert) simultaneously. Any member of the group can answer the call. Each coverage answer group is identified by a number from 1 through the maximum number allowed by your system configuration. The members of the group are identified by their extension number. Any telephone, including those administered without hardware (but not attendants) can be assigned to a coverage answer group. Example command: change coverage answer-group n, where n is the assigned group number.
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Coverage Answer Group
Ext
The extension number for each member of this coverage answer group. This number cannot be a Vector Directory Number (VDN) extension.
Group Name
The group name used to identify this group. Accepts up to 27 characters. Use the extension numbers of group members as the group name to help when determining which stations are involved in call coverage and trunk coverage paths. Example "typing pool", "room 12", "secy"
Group Number
The number associated with the Cover Answer Group.
Name
The name assigned when the member's telephone was administered.
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Chapter 33: Coverage Path
Implements Call Coverage Paths by providing the means to specify the call coverage criteria, the points in the coverage path used to redirect calls, and the number of times a principal's telephone rings before the call redirects to coverage. Example command: change coverage path n, where n is the assigned coverage path number.
Coverage Path Number
The coverage path being administered.
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Coverage Path
Cvg Enabled for VDN Route-To Party
Enables or disables the route-to party coverage path after a covered call hits a VDN vector routeto step. By default, the value is n.
Holiday Coverage
Use the Holiday Coverage field to redirect all calls during a holiday to a coverage path. For Holiday Coverage to function, set the Don't Answer field to y.
You must set the Holiday Coverage field separately for internal and external calls.
Valid Entry y
n
Usage
Communication Manager checks the Holiday Table screen for a specific holiday entry. If an entry on the Holiday Table screen matches with the current date and time, Communication Manager forwards the call to the first point that is defined in the coverage path. If there is no entry that matches with the current date and time, Communication Manager forwards the call to the subsequent point that is defined in the coverage path.
Communication Manager forwards the call to the subsequent point in the coverage path.
Holiday Table
Available only when Holiday Coverage is set to y for inside or outside calls. The number of the holiday table used for holiday coverage.
Hunt After Coverage
Valid Entry y
n
Usage
Coverage treatment continues by searching for an available station in a hunt chain that begins with the hunt-to-station assigned to the station of the last coverage point.
Coverage treatment is terminated. The call is left at the last available location, the principal or coverage point.
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Linkage
One or two additional coverage paths in the coverage path chain.
Linkage
Next Path Number
Valid Entry 1 to 9999
blank
Usage
The number of the next coverage path in a coverage path chain. If the coverage criteria of the current coverage path is dissatisfied, the system checks in this chain until it finds a coverage path with redirection criteria that matches the call status. If the chain is exhausted before the system finds a match, the call stays out of coverage.
The only path for the principal.
Criteria
Active
Use this field to assign a coverage criteria. When the coverage criteria is met, the system redirects the call to coverage.
Valid Entry y n
Usage The system redirects the call if at least one call appearance is busy. The system does not redirect the call.
Busy
Use this field to assign a coverage criteria. When the coverage criteria is met, the system redirects the call to coverage.
Valid Entry y
n
Usage
The system redirects the call if all call appearances that accept incoming calls are busy.
The system does not redirect the call.
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Coverage Path
Don't Answer
Use this field to assign a coverage criteria. When the coverage criteria is met, the system redirects the call to coverage.
Valid Entry y
n
Usage
The system redirects the call when the specified number of rings have been exceeded.
The system does not redirect the call.
All
Use this field to assign a coverage criteria. When the coverage criteria is met, the system redirects the call to coverage.
Valid Entry y
n
Usage The system redirects all calls to coverage. This option overrides any other criteria.
Calls redirect immediately to coverage. Overrides any other criteria administered for this field. The system does not redirect the call.
DND/SAC/Go to Cover
Use this field to assign a coverage criteria. When the coverage criteria is met, the system redirects the call to coverage.
Valid entry y
n
Usage
With this option, a calling user, when calling to another internal extension, can redirect a call immediately to coverage by pressing the Go to Cover button. A principal user can temporarily direct all incoming calls to coverage, regardless of the other assigned coverage criteria by pressing the Send All Calls or Do Not Disturb button. With the Send All Calls button, covering users can temporarily remove their telephones from the coverage path.
Note:
You must assign this criteria before a user can activate Do Not Disturb (Hospitality Services), Send All Calls (SAC), or Go to Cover features.
The system does not redirect the call.
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Coverage points
Logged off/PSA/TTI
Use this field to assign a coverage criteria. When the coverage criteria is met, the system redirects the call to coverage.
The system displays this field only when you set the Criteria for Logged Off/PSA/TTI Stations field to y.
Valid Entry y
n
Usage
The system redirects the call after the number of rings exceeds the value specified in the Number of Rings field. The system displays the associated Number of Rings field only when the Logged off/PSA/TTI field is set to y.
The system does not redirect the call.
Number of Rings
Valid Entry 1 to 99
Usage
The number of times a telephone rings before the system redirects the call to the first point in the coverage path. By default, the value is 2.
Coverage points
Point1, Point2, Point3, Point4, Point5, Point6
The alternate destinations that comprise a coverage path. Coverage points must be assigned sequentially without steps beginning with Point 1. Each path can have up to six coverage points. Subsequent coverage points should be unlisted if calls are redirected to:
· Message Center, a special Uniform Call Distribution hunt group · Voice messaging · The attendant These calls normally queue and never redirect to another coverage point. Calls to hunt group queue if possible. Calls redirect from a hunt group only if all hunt group members are busy and either the queue is full, or is nonexistent. If the Coverage of Calls Redirected Off-Net feature is not supported, a remote coverage point functions as the last point in the coverage path because the system can no longer control calls once they redirect off-net. However, if the Coverage of Calls Redirected Off-Net feature is enabled,
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Coverage Path
calls redirected off-net can be monitored by the system and brought back for call coverage processing.
Valid Entry extension
attd h1 to h8000 c1 to c1500 r1 to r10000 v + extension
y + extension
Usage
Redirects the call to an internal extension or announcement.
Note:
If you enter a shortened extension of the multilocation dial plan, the system does not perform certain administration and validation tasks. Therefore, the system might not display the resultant warnings or submittal denials.
Redirects the call to the attendant or attendant group. If the system has Centralized Attendant Service (CAS), the call goes to the CAS attendant.
Redirects the call to the corresponding hunt-group, for example, h32 routes to hunt group 32.
Redirects the call to the corresponding coverage answer group, for example, c20 routes to call coverage answer group 20.
Redirects the call to the corresponding remote coverage point number, for example, r27 routes to remote coverage point 27.
Redirects the call to the corresponding Vector Directory Number (VDN) extension, for example, v12345 routes to the VDN associated with extension 12345.
Note:
A VDN can be used only as the last administered point in a coverage plan.
Redirects the call to an internal extension, announcement, or the corresponding Vector Directory Number (VDN) extension as per the current date and time set in Holiday Table.
Rng
Valid Entry 1 to 99 blank
Usage
The number of rings at this coverage point before the system redirects the call to the next point in the coverage path.
Terminate to Coverage Pts. with Bridged Appearances
Valid Entry y n
Usage If activated, a call can alert as both a bridged call and a redirected call. The call skips the coverage point if it has already alerted as a bridged call.
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Chapter 34: Crisis Alert System Parameters
Defines the system parameters associated with sending crisis alert messages. Example command: change system-parameters crisis-alert
ALERT STATION
Every User Responds
Controls who needs to respond to a crisis alert.
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Crisis Alert System Parameters
Valid Entries y
n
Usage
All users who have a crisis alert button are notified and must clear the alert for every emergency alert. Crisis alert buttons should be assigned only to attendant consoles and stations that must be notified of an emergency call.
All users are notified, but only one user needs to acknowledge an alert. This user might be the attendant or any other digital telephone with a crisis alert button. When the alert is acknowledged by one user, the alert is cleared at all stations except the one that acknowledged the alert.
ALERT PAGER
Alert Pager
Supports or restricts use of the Crisis Alert to a Digital Pager.
Crisis Alert Code
The first three digits in the crisis alert pager message. This should be the numbers used to call the local emergency service or any digits used for an emergency situation (for example, 911).
Required when Crisis Alert to a Digital Pager is enabled.
Related links Alert Pager on page 180
DTMF Duration - Tone (msec)
Available only when Crisis Alert to a Digital Pager is enabled.
Valid Entry
20 to 2550 (in increments of 10)
Usage
The length of time the Dual-Tone Multi-Frequency (DTMF) tone is heard for each digit.
Related links Alert Pager on page 180
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ALERT PAGER
Main Number
Identifies the location where the crisis alert call originated. It can be the main number to the location or a numerical identification. Any dashes are for display purposes only and excluded from the message sent to the pager. This entry is the last group of digits displayed in the pager message. Accepts up to 15 digits. Available only when Crisis Alert to a Digital Pager is enabled.
Related links Alert Pager on page 180
Originating Extension
Required when Crisis Alert to a Digital Pager is enabled.
Valid Entry 1 to 16
Usage Requires a valid unassigned extension according to the dial plan
Related links Alert Pager on page 180
Pager Number
The number which Communication Manager dials to make the call to the pager. The ARS access code is automatically pre-pended to the Pager Number before Communication Manager dials the call.
Pause (msec)
Available when Crisis Alert to a Digital Pager is enabled.
Valid Entry
20 to 2550 (in increments of 10)
Usage The length of time between DTMF tones for each digit.
Related links Alert Pager on page 180
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Crisis Alert System Parameters
Pin Number
If the pager service requires you to submit a PIN number, type the number in this field. The PIN number can be up to 15 digits. You can also enter the p (pause), # and * characters. A pause of up to 2 seconds can be used for the timing of the message. For instance, a pause might be necessary to allow time for the pager service to set up the correct pager message box.
The system displays this field only if Crisis Alert to a Digital Pager is enabled.
Related links Alert Pager on page 180
Retries
Available when Crisis Alert to A Digital Pager is enabled.
Valid Entry 0 to 10
Usage
The number of times the system tries to send out the alert message in case of an unsuccessful attempt. This increases the chances that the pager receives a crisis alert message.
Related links Alert Pager on page 180
Retry Interval (sec)
Available when the Crisis Alert to a Digital Pager is enabled and Retries is set to a value from 1 to 10.
Valid Entry 30 to 60
Usage
The time in seconds between retries. If an attempt to call the pager fails, the retry call attempts after the retry interval period.
Related links Alert Pager on page 180 Retries on page 182
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SNMP NOTIFICATION
SNMP NOTIFICATION
SNMP Inform to Notify Adjunct When DCP and H.323 Stations Go In-Service
Enables or disables sending SNMP informs to a trap receiver when a DCP station comes into service, or an H.323 station is registered.
Valid Entries y
n
Usage
Communication Manager sends SNMP informs to a trap receiver when a DCP station comes into service, or an H.323 station is registered.
Communication Manager does not send SNMP informs to a trap receiver when a DCP station comes into service, or an H.323 station is registered. This is the default value.
SNMP Inform to Notify Adjunct When SIP Station Dials Emergency Call
Controls sending inform traps to SNMP receiver when a SIP station places an emergency call.
Note:
Ensure that, in the FP Traps screen of the Communication Manager System Management Interface (SMI), you set the Notification field to Inform for SNMP v3
Valid Entries y
n
Usage
Communication Manager sends inform traps to SNMP receiver when a SIP station places an emergency call.
Communication Manager does not send inform traps to SNMP receiver when a SIP station places an emergency call. This is the default value.
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Chapter 35: CTI Link
Available if either ASAI Link Core Capabilities or Computer Telephony Adjunct Links are enabled on the system. Example command: add cti-link n, where n is the CTI link number. Related links
ASAI Link Core Capabilities on page 829 Computer Telephony Adjunct Links on page 831
CTI link: page 1
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CTI link: page 1
COR
Class of Restriction (COR) number with the required restriction.
CTI Link
The Computer Telephony Integration (CTI) link number.
Extension
The extension for this link.
Name
The name associated with this CTI link.
Type
Use this field to specify the type of Computer Telephony Integration (CTI) link.
Valid entry ADJ-IP ASAI-IP
Usage CTI links administered without hardware. CTI links administered without hardware.
The associated adjunct controls agent logins, and the logged-in agents can use the data terminal keyboards to perform telephone functions.
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CTI Link
CTI link: page 2
Block CMS Move Agent Events
Enables or disables the blocking of certain event report messages involved with the move of agents while staffed. If the Call Management System sends an agent-move-while-staffed message (MVAGSFD8), ASAI does not send the associated agent Logout Event Report (C_Logout), Login Event Report (C_login) and Agent Work Mode Change event report messages.
Event Minimization
Enables or disables event minimization for this link that limits the number of event reports sent to an adjunct. This option can be used when event reports normally would be sent on multiple associations, but the adjunct can see only a single report. Typically, these event reports are identical except for the association they are sent over (for example, call control, domain control, or active notification). Some applications discard duplicate events, so in this case, there is no point in sending them across the ASAI CTI link. When activated, only a single such event can be sent. The selection of the association on which the event is sent is based on association precedence as follows: active notification (if enabled), call control (if enabled), or domain control (if enabled).
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CTI link: page 2
IC Adjunct Routing
IC Adjunct Routing enables or disables the Enhanced Adjunct Routing wait sequence. When disabled, the wait sequence consists of only one wait or announcement step following one or more adjunct-route steps. When enabled, the wait sequence can contain multiple wait steps, announcement steps, goto steps, and the collect step. If the adjunct route step is successful, the adjunct must respond before the wait sequence completes. Otherwise, the adjunct route step is cancelled. If the adjunct route step fails, the vector steps in the wait sequence are skipped.
Remove '+' from SIP Numbers
If you set the Remove '+' from SIP Numbers field on the CTI Link screen to y and if a leading plus (+) sign is present in a calling or called number sent over a CTI link, Communication Manager removes the leading plus (+) sign from the number. The default value is y.
Send Disconnect Event for Bridged Appearance
Indicates whether or not an event report is sent when a bridged appearance disconnects.
Special Character for Restricted Number
Enables or disables an ASAI CTI link that indicates the calling number restricted presentation within an event report. When enabled, a calling number received in a SETUP message has the presentation indicator set (octet 3a in the calling number), then "*" is appended to the calling party number in the ASAI message.
Two-Digit Aux Work Reason Codes
Use this field to send a two-digit Reason code over the ASAI link. All messages that include the Aux-work Reason code allow codes from 1 to 99. Related links Two-Digit Aux Work Reason Codes on page 346
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CTI Link
CTI link: page 3
Bridged Appearance Origination Restriction
Restricts or allows call origination on the bridged appearance.
Valid Entry y n
Usage
Call origination on the bridged appearance is restricted.
Call origination ion the bridged appearance is allowed. This is normal behavior, and is the default.
SAC/CF Override
With SAC/CF Override, the user of the calling station can override the redirection set by the called station.
Valid entry ask
Usage
The system prompts the user of the calling station whether the call must follow the redirection path or override the redirection path. The user can type y or n.
Table continues...
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Valid entry no
yes
CTI link: page 3
Usage The user of the calling station cannot override the redirection path of the call. The call follows the redirection path. The user of the calling station can override the redirection path of the call, provided the called station has at least one idle call appearance.
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Chapter 36: Data Module
Example command: change data-module n, where n is the module number.
Data module: page 1
BCC
Indicates the value that corresponds to the speed setting of the data module. Used with Data Line, Netcon, Processor Interface, Point-to-Point Protocol, Processor/Trunk (pdm selection), and System Port Data Modules. This field can be compared with the BCC value in an associated routing pattern when attempted calls utilizing the data module fail to complete. The BCC values must be the same.
Available only when ISDN-PRI or ISDN-BRI trunks are enabled.
Valid Entry 1 2, 3, 4
Usage Relates to 56-bkps Relates to 64 kbps
Related links ISDN-PRI on page 836 ISDN-BRI Trunks on page 835
Connected to
Used with Data Line and Processor/Trunk (pdm selection) Data Modules.
Valid Entry dte isn
Usage The Asynchronous Data Unit (ADU) is connected to Data Terminal Equipment. The ADU is connected to the Information Systems Network.
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Data module: page 1
COS
The system does not display this field for ethernet.
Valid entry 0 to 15
Usage The COS for the data module
COR
The system does not display this field for ethernet.
Valid entry 0 to 995
Usage The COR for the data module.
Data Extension
Use this field to assign an extension to the data module. You can enter up to five digits. This value must conform to the dial plan.
ITC
The Information Transfer Capability (ITC) is used with 7500, Announcement, data-line, Netcon, Processor/ Trunk (pdm selection), Processor Interface, and System Port Data Modules. Indicates the type of transmission facilities used for ISDN calls originating from this endpoint. Unavailable for voice-only or BRI stations.
Valid Entry restricted
unrestricted
Usage
Either restricted or unrestricted transmission facilities are used to complete the call. A restricted facility is a transmission facility that enforces 1's density digital transmission. In other words, a sequence of eight digital zeros is converted to a sequence of seven zeros and a digital one.
Only unrestricted transmission facilities are used to complete the call. An unrestricted facility is a transmission facility that does not enforce 1's density digital transmission. In other words, digital information is sent exactly as is.
Name
The name of the user associated with the data module. The name is optional and can be blank. Accepts up to 27 alphanumeric characters.
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Data Module
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Port
A port location to which the data module is connected. Used with 7500, Data Line, Ethernet, Processor/Trunk, PPP, System Port, and World Class BRI Data Modules.
Characters 1 to 2 3 4 to 5 6 to 7
Meaning Cabinet number Carrier Slot number Circuit Number
Value 1 to 64 A to E 0 to 20 · 01 to 31 · 01 to 16 · 01 to 08
17/33
Note:
An x in the Port field indicates that there is no hardware associated with the port assignment, also known as Administration Without Hardware (AWOH). These stations are called phantom stations. If this data module is designated as a secondary data module, you cannot enter x. The port of a primary data module cannot be changed to x if a secondary data module is administered.
Remote Loop-Around Test
Used with Processor/Trunk Data Modules. Available with a pdm or tdm type trunk.
Valid Entry y
n
Usage
The data module supports a loop-back test at the EIA interface. Avaya equipment supports this loop-back test which is not required by Level 2 Digital Communications Protocol.
Abort a request for this test
Related links Type on page 193
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Data module: page 1
Secondary data module
Used with Processor/Trunk Data Modules. Available with a pdm type trunk. The primary data module must be administered before the secondary data module can be added. A data module administered without port hardware cannot be the secondary data module.
Valid Entry y n
Usage This PDM is the secondary data module used for Dual I-channel AUDIX networking. This is the primary PDM, or this data module is not used for AUDIX networking.
Related links Type on page 193 Port on page 192
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
Type
The type of data module.
Valid Entry 7500
announcement
Usage
Assigns a 7500 Data Module. The 7500 data module supports automatic TEI, Bchannel, maintenance and management messaging, and SPID initialization capabilities. Both voice and data BRI endpoints are assigned to either the ISDN-BRI - 4-wire S/T-NT Interface circuit pack or the ISDN-BRI - 2-wire U circuit pack. Each can support up to 12 ports. Since BRI provides multipoint capability, more than one ISDN endpoint (voice or data) can be administered on one port. For BRI, telephones having SPID initialization capabilities support multipoint administration. Multipoint administration can only be activated if endpoints administered on the same port is not a fixed tie endpoint and station on the same port does not have B-channel data capability. Currently, multipoint is restricted to two endpoints per port.
Assigns an announcement data module. The announcement data module is built-in to the integrated announcement circuit pack. The system uses this data module to save and restore the recorded announcements file between the announcement circuit pack and the system memory.
Table continues...
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Data Module Valid Entry data-line
ethernet ni-bri pdm
Usage
Assigns a Data Line Data Module. Administered ports on the Data Line circuit pack (DLC) can connect EIA 232C devices to the system. The DLC, with a companion Asynchronous Data Unit (ADU), provides a less expensive data interface to the system than other asynchronous DCP data modules.
The DLC supports asynchronous transmissions at speeds of Low and 300, 1200, 2400, 4800, 9600, and 19200 bps over two-pair full-duplex lines. These lines can have different lengths, depending on the transmission speed and wire gauge.
The DLC has eight ports. The connection from the port to the EIA device is direct, meaning that no multiplexing is involved. A single port of the DLC is equivalent in functionality to a data module and a digital line port. The DLC is displayed as a data module to the Digital Terminal Equipment (DTE) and as a digital line port to the server running Communication Manager.
The DLC connects the following EIA 232C equipment to the system:
· printers
· non-intelligent data terminals
· intelligent terminals, personal computers (PCs)
· host computers
· Information Systems Network (ISN), RS-232C local area networks (LANs), or other data switches
Assigns an Ethernet data module. A 10BaseT port can be administered on the Control-LAN (C Lan) circuit pack. This port provides a TCP/IP connection to network hub or LAN.
Assigns an NI-BRI Data Module.
Assigns a DCE interface for Processor/Trunk Data Modules. A Modular Processor Data Modules (MPDMs) and Modular Trunk Data Modules (MTDMs) can be administered.
The MPDM, 7400B, or 8400B Data Module provides a Data Communications Equipment (DCE) interface for connection to equipment such as data terminals, CDR output devices, on-premises administration terminal, Message Server, Property Management System (PMS), AUDIX, and host computers. It also provides a Digital Communications Protocol (DCP) interface to the digital switch. DCE is the equipment on the network side of a communications link that provides all the functions required to make the binary serial data from the source or transmitter compatible with the communications channel.
The MTDM provides an Electronic Industries Association (EIA) Data Terminal Equipment (DTE) interface for connection to off-premises private line trunk facilities or a switched telecommunications network and a DCP interface for connection to the digital switch. DTE is the equipment comprising the endpoints in a connection over a data circuit. For example, in a connection between a data terminal and a host computer, the terminal, the host, and their associated modems or data modules make up the DTE. The MTDM or 7400A Data Module also can serve as part of a conversion resource for Combined Modem Pooling.
Table continues...
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Valid Entry ppp
system-port tdm wcbri
Data module: page 1
Usage Assigns a Point-to-Point Protocol data module. A synchronous TCP/IP port can be administered on the C-Lan circuit pack. These ports are tailored to provide TCP/IP connections for use over telephone lines. Assigns a System Port Data Module. Assigns a DTE interface for Processor/Trunk Data Modules. See the pdm entry above. Assigns a World Class BRI Data Module.
ABBREVIATED DIALING
List1
The abbreviated dialing list for the data module. Used with 7500, Data Line, Netcon, Processor/ Trunk, Processor Interface, and World Class BRI Data Modules. Supports Data Hot Line. This field can be left blank.
Valid Entry e g p s
Usage Enhanced Requires a group list number. Requires a personal list number. System
ASSIGNED MEMBER
Ext and Name
The extension number and name of the previously administered user with associated Data Extension buttons, who shares the module. Used with Data Line, Announcement, Netcon, Processor/Trunk, Processor Interface, and System Port Data Modules.
SPECIAL DIALING OPTION
Identifies the type of dialing for calls when this data module originates calls. Used with 7500, Data Line, Netcon, Processor/Trunk, Processor Interface, and World Class BRI Data Modules.
Valid Entry hot-line
Usage
When the user goes off-hook on the data module, the hot line destination number gets dialed.
Table continues...
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Data Module
Valid Entry default blank
Usage
When the user goes off-hook on the data module and presses Enter at the DIAL prompt, the default dialing destination number gets dialed.
Normal keyboard dialing
Data module: page 2: Type data-line
CAPABILITIES
Busy Out
Enables or disables the placement of the DLC port in a busied-out state once the DTE control lead to the DLC is dropped. This option should be enabled for DTEs that are members of a hunt group and to activate busy out when DTE is turned off, so that calls do not terminate on that DTE.
Configuration
Activates or deactivates the viewing and changing of options from the DTE. Available only when KYBD Dialing is enabled. This option normally is enabled for originate/ receive DTE such as nonintelligent terminals and disabled for intelligent devices such as computers.
Related links KYBD Dialing on page 196
KYBD Dialing
Note:
ADU-type hunt groups connecting the system to terminal servers on a host computer should have these hunt group extensions assigned as no keyboard dialing.
Valid Entry y
n
Usage
Enables keyboard dialing, allowing data endpoints to originate calls using the EIA 232C interface and obtain ASCII feedback text. The user gets the dial prompt. This option normally is enabled for a originate/receive DTE that has a need to set up data calls.
Disables keyboard dialing. Calls do not originate at the DTE and the system does not display the text feedback during call setup or cancellation. Data call answering is still supported but without text feedback. If the Low speed setting is enabled, keyboard dialing should be disabled.
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Data module: page 2: Type data-line
OPTIONS
Answer Text
Available only with KYBD Dialing.
Applies to the following messages:
· INCOMING CALL · ANSWERED · DISCONNECTED · DISCONNECTED OTHER END
Valid Entry y
n
Usage
Text messages can be delivered to the DTE when a call is being answered. Applies to DLC-generated text as well as text received from the system.
The system still generates the text, but the DLC prevents it from being sent to the device. Usually is disabled when the answering DTE is a computer or an intelligent device.
Related links KYBD Dialing on page 196
Connected Indication
Available only with KYBD Dialing.
Valid Entry y
n
Usage
Generates a "CONNECTED" message to the DTE when the connection has been established.
The connected indication is provided by the DLC activating its EIA 232C control lead.
Related links KYBD Dialing on page 196
Dial Echoing
Available only with KYBD Dialing.
Valid Entry y n
Usage
Echos characters back to the DTE.
Disables Dial Echoing. Dial Echoing should be disabled when keyboard dialing is done by an intelligent device.
Related links KYBD Dialing on page 196
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Data Module
Disconnect Sequence
Selects the sequence for a disconnect. Available only with KYBD Dialing.
Valid Entry long-break two-breaks
Usage A long-break is greater than 2 seconds. Two-breaks is within 1 second.
Related links KYBD Dialing on page 196
Parity
Valid Entry even odd mark space
Usage
The type of parity. The DLC generates the parities when call setup text is sent to the DTE. The DLC ignores the parity when receiving dialing characters. Parity has no bearing on the far end. It is used by the DLC-to-terminal communications during call setup. Available only with KYBD Dialing.
Related links KYBD Dialing on page 196
Permit Mismatch
Enables or disables the Permit Mismatch feature. Permit Mismatch:
· The EIA interface can operate at a rate different than that agreed to in the data module handshake. The data module handshake is always the highest compatible rate as determined by the reported speed option of each data module.
· Instructs the DLC to operate at the highest selected speed, which is a higher rate than the far-end data module. The DLC reports the highest-optioned speed, all the lower speeds, or the previously-selected auto-adjust speed during the handshake process.
· Eliminates the need to change the DTE/DLC speed every time a call is placed to and from an endpoint operating at a different speed.
Caution:
You must be cautious while sending information from a DTE/ DCE that is transmitting data at higher rates than that of the far end. Sustained usage of this type of transmission might result in data loss. When you enable this option, the DTE must match the highest speed selected for the associated DLC port.
This option is meant for use by a DTE device operating locally at a higher baud rate than that of its far-end connection, but transmitting relatively low amounts of data, for example, a user typing at a terminal.
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Data module: page 2 - Type 7500, WC-BRI, NI-BRI
Note: The Low speed setting is not reported as an available speed when Permit Mismatch is enabled. Related links SPEEDS on page 199
SPEEDS
Enables or disables the following operating speeds:
Valid Entry Low
300, 1200, 2400, 4800, 9600, or 19200
Autoadjust
Usage
Instructs the DLC to operate at a low speed from 0 to 1800 bits per second (bps). Disable if KYBD Dialing is enabled.
The DLC can be any one of these speeds. The speed is matched for the duration of the call, from call setup to call takedown. For multiple speeds, select three or more entries. When you select multiple speeds and disable autoadjust, the DTE speed must be the highest selected speed. This is required because all feedback text is delivered to the DTE at the highest selected speed.
· Tells the DLC port to automatically adjust to the operating speed and to the parity of the DTE it is connected to
· Can be used with any of the speeds listed
· Applies only to calls originated by the user through Keyboard Dialing
· Available only when KYBD Dialing is enabled
Related links KYBD Dialing on page 196
Data module: page 2 - Type 7500, WC-BRI, NI-BRI
BRI LINK/MAINTENANCE PARAMETERS
Endpt Init
Indicates whether the terminal endpoint has initialization capability. Endpoint initialization is a procedure required for multipoint operation, by which User Service Order Profile (USOP) is associated with an endpoint on the ISDN-BRI. This association is made through a Service Profile Identifier (SPID) administered into the system and entered into the ISDN-BRI terminal. For a ISDN-BRI terminal to become operational in a multipoint configuration, both the administered
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Data Module
SPID and the SPID programmed into the ISDN-BRI terminal must be the same. This means that the SPID of the new or reused terminals must be programmed to match the administered SPID value. Used with 7500, World Class BRI, and NI-BRI Data Modules.
Fixed TEI
Indicates whether an endpoint has Fixed Terminal Equipment Identifier (TEI) capability. TEI identifies a unique access point within a service. For Fixed TEI stations, the TEI must be administered. For terminals with automatic TEI capability, the associated TEI is assigned by the system. Used with 7500, World Class BRI, and NI-BRI Data Modules.
MIM Mtce/Mgt
Enables or disables Management Information Message (MIM) support. MIM provides terminal support for MIM Maintenance and Management capabilities, other than endpoint initialization. Used with 7500 Data Modules.
MIM Support
Enables or disables the capability of MIM endpoint initialization (SPID support), and other Maintenance/Management. Used with 7500 Data Modules.
SPID
The Service Profile Identifier (SPID) is a variable parameter of up to 10 digits. The SPID must be different for all terminals on the ISDN-BRI and from the Service SPID. The SPID should always be assigned. If the SPID is unassigned for the first ISDN-BRI on a port, any other ISDN-BRI assignment to that port is blocked. Used with 7500, World Class BRI, and NI-BRI Data Modules. Available only if Endpt Init is enabled.
Related links Endpt Init on page 199
TEI
Available only if Fixed TEI is enabled.
Valid entry 0 to 63
Usage
The Terminal Endpoint Identifier (TEI) is a layer 2 addressing parameter used by Communication Manager to exchange information with BRI endpoints over the pointto-point signaling link. Used with 7500, World Class BRI, and NI-BRI Data Modules.
Related links Fixed TEI on page 200
XID
Enables or disables layer 2 Exchange identification (XID) testing capability. Used with 7500, World Class BRI, and NI-BRI Data Modules. In almost all cases, disable the field.
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Chapter 37: Date and Time
Displays the system date and time, selects the daylight saving plan number, if any, and shows whether the current time is standard time or daylight saving. Settings on this screen affect the internal clock and timestamp of the server running Communication Manager. Update the date and time for a leap year or a system restart after a power failure. The correct date and time assure that CDR records are correct. CDR does not work until you enter the date and time. Example command: set time
Day of the Month
Valid entry 1 to 31
Usage The current day of the month. The system clock uses this as the current date.
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Date and Time
Day of the Week
Valid Entry
Sunday through Saturday
Usage The current day of the week. The system clock uses this as the current day.
Daylight Saving Rule
Valid Entry 0 to 15
Usage
The daylight saving rule in use for the system. The system clock uses this as the current daylight saving rule.
Related links Daylight Saving Rules on page 204
Hour
Valid Entry 0 to 23
Usage
The current hour to be used by the system clock. The system uses a 24-hour clock. For example, 14:00 is the same as 2:00 p.m.
Minute
Valid Entry 0 to 59
Usage The current minute. The system clock uses this as the current minute.
Month
Valid Entry January to December
Usage The current month. The system clock uses this as the current month.
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Second
Displays the seconds and cannot be modified. Resets to zero when saved.
Type
Valid Entry daylight-savings standard
Usage Indicates daylight saving time is in effect. Indicates standard time is in effect.
Year
The current year in 20XX format. The system clock uses this as the current year.
Second
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Chapter 38: Daylight Saving Rules
Administers up to 15 customized daylight saving rules. Specifies the exact date and time each daylight saving rule goes into effect and when it stops. Rule 0 makes no adjustment to the system clock for daylight saving and cannot be modified. In this example, Rule 1 applies to all time zones in the U.S. and begins on the first Sunday on or after March 8 at 2:00 a.m. with a 01:00 increment. Daylight Saving Time stops on the first Sunday on or after November 1 at 2:00 a.m., also with a 01:00 increment used as a decrement when switching back to Standard time. Telephone displays reflect these settings.
1: Start: first _ Sunday_ on or after _March 8_ at 2:00_ 01:00__ Stop: first _ Sunday_ on or after November 1_ at 2:00_
Example command: change daylight-savings-rules
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Change Day (Start)
Change Day (Start)
Valid Entry
Sunday to Saturday
Day
Usage The day of the week the clock moves ahead to begin daylight saving.
The clock changes on the exact date entered for Month (Start) and Date (Start) values.
Related links Month (Start) on page 206
Change Day (Stop)
Valid Entry
Sunday to Saturday
Day
Usage The day of the week the clock moves back to standard time
The clock changes on the exact date entered for the Month (Stop) and Date (Stop) values
Related links Month (Stop) on page 206
Date (Start)
Valid Entry 0 to 31
Usage The day of the month the clock moves ahead to begin daylight saving.
Date (Stop)
Valid Entry 0 to 31
Usage The day of the month the clock moves back to standard time
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Daylight Saving Rules
Increment (Start)
Valid Entry 0 to 23
0 to 59
Usage
The number of hours the clock moves ahead for daylight saving and moves back to return to standard time.
The number of minutes you want the clock to move ahead for daylight saving and to move back to return to standard time.
Month (Start)
Valid Entry
January to December
Usage The month the clock moves ahead to begin daylight savings
Month (Stop)
Valid Entry
January to December
Usage The month the clock moves back to standard time
Rule
The daylight saving rule number.
Time (Start)
Valid Entry 0 to 23
0 to 59
Usage
The hour the clock moves ahead to begin daylight saving. The system uses a 24hour clock. For example, 14:00 is the same as 2:00 p.m.
The minute the clock moves ahead to begin daylight saving.
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Time (Stop)
Time (Stop)
Valid Entry 0 to 23
0 to 59
Usage
The hour the clock moves back to standard time. The system uses a 24-hour clock. For example, 14:00 is the same as 2:00 p.m.
The minute the clock moves back to standard time
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Chapter 39: DCS to QSIG TSC Gateway
Determines when and how to convert messages from an administered AUDIX NCA-TSC to a QSIG NCA-TSC. Maps the AUDIX NCA-TSC to the appropriate machine ID index to find the QSIG subscriber entry in the QSIG MWI-Prefix screen. Assigns the voice mail number used when a DCS served-user node interrogates a QSIG message center. Available only if Interworking with DCS is enabled for the system. Example command: change isdn dcs-qsig-tsc-gateway Related links
Interworking with DCS on page 849
AAR/ARS Access Code
The AAR/ARS Access Code. Accepts up to four digits, including characters * and #.
Mach ID
Valid Entry digits
Usage
A unique machine ID. Do not repeat a machine ID if it is already associated with a processor channel on an Ethernet link. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Sig Grp
Valid Entry digits
Usage
The assigned signaling group number for each machine ID. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
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TSC Index
TSC Index
The TSC Index for each machine ID.
Valid Entry 1 to 64
Usage The assigned signaling group number for the qsig-mwi application type.
Voice Mail Number
The complete Voice Mail Dial Up number. Accepts up to 17 digits.
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Chapter 40: Dial Plan Analysis Table
The Dial Plan Analysis Table is the system's guide to translating the digits dialed by users. It determines the beginning digits and total length for each type of call that Communication Manager needs to interpret. The Dial Plan Analysis Table and the Dial Plan Parameters screen work together to define the system's dial plan. Example command: change dialplan analysis
Call Type
Valid Entry aar ars
attd
dac enb-ext
ext
fac pext
udp
Usage
Automatic Alternate Routing -- Routes calls within a company's own private network. Requires that ARS/ AAR Dialing without FAC is enabled.
Automatic Route Selection -- Routes calls that go outside a company over public networks. ARS also routes calls to remote company locations without a private network. Requires that ARS/ AAR Dialing without FAC is enabled.
Attendant -- Defines how users call an attendant. If a telephone's COR restricts the user from originating calls, this user cannot access the attendant using this code. The attendant access code can also be administered by entering an fac or dac.
Dial access code -- Using this feature, you can use trunk access codes (TAC) and feature access codes (FAC) in the same range.
Enbloc extension -- Defines a block of extensions that must be dialed using a prefix when the caller dials from a keypad. These extensions can be dialed without a prefix if the caller dials enbloc, for example, from a station call log.
Primary extension -- Defines extension ranges that can be used on the system. Extension can have a first digit of 0 through 9 and can have up to 10 digits. Extension cannot have the same first digit as a 1-digit ARS or AAR feature access code (FAC).
Feature access code only -- Users dial an FAC instead of programming a button.
Prefixed extension -- Identifies the call type as an extension. After digit collection, the prefix digit is removed from the string of dialed digits. The remaining digits make up the extension number and are then processed. With a prefixed extension, extensions numbers with any dialed string can be used. The extension length must be specified on the table.
Uniform Dial Plan -- Shares a common dial plan among a group of servers.
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Related links ARS/AAR Dialing without FAC on page 829 Dial Access on page 507 UDP Extension Search Order on page 217
Dialed String
Dialed String
The digits that Communication Manager analyzes to determine how to process the call. Two Dial Plan entries can use the same Dialed String only if the Dialed String consists of one digit. Longer Dialed Strings must all be unique. A new entry cannot be administered if it causes an existing extension, feature access code, or trunk access code to become inaccessible.
Call Type aar ars attd dac
enb-ext ext
fac
Valid Entry 0 to 9 0 to 9 0 to 9 0 to 9 * # 0 to 9
0 to 9
0 to 9 * #
Usage Dialed String entry can be one to six digits long. Dialed String entry can be one to six digits long. Dialed String entry can be one or two digits long. Dialed String can be one to four characters long.
The characters "*" and "#" can only be used as a first digit.
Dialed String can be one to ten digits long. The corresponding Total Length can be up to sixteen digits. Dialed String can be one to ten digits long.
The extension cannot have the same first digit as a one-digit ARS or AAR feature access code (FAC).
The corresponding Total Length can be up to sixteen digits. Dialed String can be one to four characters long.
The characters "*" and "#" can only be used as a first digit.
A FAC must have the longest total length for a given dialed string when using mixed numbering. Otherwise, problems might occur when, for example, three-digit FACs and four-digit extensions begin with the same first digit and the FAC is an abbreviated dialing list access code.
However, if the entry in the dial plan that defines the FAC is used to define the AAR or ARS access code, then it must have the longest total length in the dial plan.
Table continues...
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Dial Plan Analysis Table
Call Type pext
Valid Entry 0 to 9
udp
0 to 9
Usage
The prefix is the first digit and an extension number of up to five digits in length.
The maximum length of a prefix and extension combination is six digits long.
A prefixed extension cannot have the same first digit as a dial access code.
A prefixed extension cannot have the same dialed string as the ARS or AAR FAC.
Dialed String can be one to ten digits long.
The corresponding Total Length can be up to eighteen digits.
Location
This field displays the location of the telephone that is used to dial the number. For more
information about assigning a location and for a list of features that use the location, see Avaya Aura® Communication Manager Feature Description and Implementation.
Valid entry
1 to 250 or 1 to 2000
all
Usage
If there are matching entries in the location of the telephone, the system displays the matched entry.
If there are no matching entries in the location of the telephone, the system displays all.
Related links Multiple Locations on page 838
Percent Full
Valid Entry 0 to 100
Usage
The percentage of system memory resources that have been allocated for the dial plan currently used. For details on the system memory resources, see Avaya Aura® Communication Manager System Capacities Table.
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Total Length
Total Length
The number of digits for this call type. Must be greater than or equal to the number of digits in the Dialed String.
Valid Entry 1 to 2 1 to 4 1 to 4 2 to 6 1 to 16 1 to 18
Usage attd dac fac pext enb-ext udp
Related links Dialed String on page 211
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Chapter 41: Dial Plan Parameters
The Dial Plan Parameters screen works with the Dial Plan Analysis Table to define the system dial plan. It also controls the appearance of digit extensions on station displays. These multi-digit extensions can be hard to read as a block. Using Communication Manager you can select the display format for 6- to 16-digit extensions.
AAR/ARS Internal Call Prefix
The digits entered here get concatenated with the calling or called extension. Accepts up to eight digits that excludes * or #. Available only when ARS/AAR Dialing Without FAC is enabled for the system. Requires administration of AAR/ARS Internal Call Total Length.
Related links AAR/ARS Internal Call Total Length on page 214
AAR/ARS Internal Call Total Length
Available only if ARS/AAR Dialing Without FAC is enabled for the system.
Valid Entry 6 to 10 blank
Usage
The total length of the internal call digit string, including the Internal Call Prefix and the calling or called extension. Requires administration of an AAR/ARS Internal Call Prefix.
Note:
The longest extension length on the Dial Plan Analysis table, plus the length of the ARS/AAR Internal Call Prefix, must be equal to or greater than the ARS/AAR Internal Call Total Length value.
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Call Type Analysis Behavior When Extension Not Assigned
Call Type Analysis Behavior When Extension Not Assigned
Use this field to specify how Communication Manager must route a call when: · The dialed string matches an ext or udp entry in the Call Type Analysis table. · The resulting extension is valid, but unassigned.
Note:
You can define valid extensions by assigning the network extensions to a block of numbers in the Uniform Dial Plan table.
Valid entry intercepttreatment
try-nextchoice
Usage
Communication Manager stops processing the digits and routes the call to the appropriate intercept treatment:
· For a local call, the intercept treatment is defined in the Invalid Number Dialed Intercept Treatment field.
· For a trunk call, the intercept treatment is defined in the DID/Tie/ISDN/SIP Intercept Treatment field.
Communication Manager processes the next administered choice. If the next choice is not administered, Communication Manager routes the call using the original digits. The default option is try-next-choice.
ETA Node Number
Valid Entry 1 to 999 blank
Usage
The number of the destination server for Extended Trunk Access (ETA) calls. ETA calls are unrecognized numbers you can send to another switch for analysis and routing. Such numbers can be Facility Access Codes, Trunk Access Codes, or extensions outside the UDP table.
ETA Routing Pattern
Valid Entry 1 to 999
Usage The number of the routing pattern to reach the destination server.
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Dial Plan Parameters
Local Node Number
Valid Entry 1 to 63
blank
Usage
Identifies a specific node in a server network. Must match the DCS switch node number and the CDR node number if they are specified.
Leave blank if automatic restoration, DCS, and CDR are unused.
Retry ARS/AAR Analysis if All-Location Entry Inaccessible
Available with Communication Manager Release 4.0.x or later.
Valid Entry y
n
Usage
The system finds and uses the best possible entry in the per-location ARS table, if the all-location table points to a trunk group that cannot be accessed because the network has fragmented.
The system stops retry of ARS analysis when a trunk group cannot be accessed as a result of the network being fragmented.
UDP-ARS Calls Considered Offnet
Valid entry y
n
Usage
If you set the Net field to ARS and the UDP-ARS Calls Considered Offnet field to y, the call is offnet.
If UDP-ARS calls are off-net then:
· Users with the COS setting Restrict Call Fwd-Off Net enabled cannot forward calls to a UDP-ARS number.
· Stations that are outward-restricted (through the COR) cannot place calls to a UDP-ARS number.
· Toll analysis is applied when a call is placed by a toll-restricted user to a UDP-ARS number.
· COR-COR checks are inapplicable when a user dials a call to a UDP-ARS number.
If you set the Net field to ARS and the UDP-ARS Calls Considered Offnet field to n, the call is onnet.
For more information about the Net field, see Uniform Dial Plan Table section of the Avaya Aura® Communication Manager Screen Reference.
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UDP Extension Search Order
UDP Extension Search Order
Specifies the first table to search to match a dialed extension. If the dialed extension is not found in the specified place, the system searches for the dialed extension in the alternate place. Available if Uniform Dialing Plan is enabled for the system.
Valid Entry
local-extensionsfirst
udp-table-first
Usage
Searches the local server first to match a dialed extension, and if not found, then uses the UDP tables to route the call.
Searches the UDP tables for an off-switch conversion, and if not found, then searches the local server for the dialed extension.
EXTENSION DISPLAY FORMATS
Extension display format
Specifies how the system punctuates extensions for display. The punctuation field is divided into two columns, one for Inter-Location/SAT displays, and one for Intra-Location displays.
In Communication Manager 6.0, the Inter-Location/SAT column is divided into two columns. You can insert a dial prefix in the Inter-Location column, if appropriate, before an enbloc extension. With this, called stations can store a dialable number in their call logs.
Note:
The maximum length of a displayed extension (including punctuation) varies based on the extension length -- for example, for a 10digit extension, the maximum punctuated length is 13 characters. You must trade off a punctuation mark so that you can insert a prefix digit. You cannot insert a prefix digit or a punctuation mark into the inter-location format for an extension of length 13,14,15, or 16.
Blank spaces are sometimes used in telephone extensions, especially outside of the U.S. Dots (.) are used on SAT screens in place of blanks. The following table gives the maximum number of punctuation marks permitted for each extension length.
The number of punctuation marks that the system supports is determined by the number of "x"s in the format.
· If the format contains fewer than six x's, no punctuation marks can be entered.
· If the format contains six or more x's, the maximum number of punctuation marks is determined by the following table.
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Dial Plan Parameters
Extension Length 6 7 8 9 10 11 12 13 14 15 16
Maximum Punctuation Marks 2 1 3 3 3 2 1 0 0 0 0
Maximum Total Length 8 8 11 12 13 13 13 13 14 15 16
Valid Entry xx.xx.xx
Usage
Can contain all "x" characters (no punctuation) or a combination of "x" characters and 0 to 2 hyphens (-), spaces, or periods (.) to depict how extensions display. A format must be specified or the default takes effect. This field cannot be left blank. The default values for the 8-, 9-, 10-, 11-, 12-, 13- and 16-digit fields are shown in the following example.
EXTENSION DISPLAY FORMATS 6-Digit Extension: 7-Digit Extension: 8-Digit Extension: 9-Digit Extension:
10-Digit Extension: 11-Digit Extension: 12-Digit Extension: 13-Digit Extension: 14-Digit Extension: 15-Digit Extension: 16-Digit Extension:
SAT xx.xx.xx xxx-xxxx xx.xx.xx.xx xxx-xxx-xxx xxx-xxx-xxxx xxxx-xxx-xxxx xxxxxx-xxxxxx xxxxxxxxxxxxx xxxxxxxxxxxxxx xxxxxxxxxxxxxxx xxxxxxxxxxxxxxxx
Inter-Location xx.xx.xx xxx-xxxx xx.xx.xx.xx xxx-xxx-xxx xxx-xxx-xxxx xxxx-xxx-xxxx xxxxxx-xxxxxx xxxxxxxxxxxxx xxxxxxxxxxxxxx xxxxxxxxxxxxxxx xxxxxxxxxxxxxxxx
Intra-Location xx.xx.xx xxx-xxxx xx.xx.xx.xx xxx-xxx-xxx xxx-xxx-xxxx xxxx-xxx-xxxx xxxxxx-xxxxxx xxxxxxxxxxxxx xxxxxxxxxxxxxx xxxxxxxxxxxxxxx xxxxxxxxxxxxxxxx
Location-Based Call Type Analysis
Use this field to enable the user to make a call to an internal contact using a short or a prefixed extension.
When this field is enabled and the user dials the short digit extension of an internal contact using the dialpad or call logs or contact from a supported endpoint, Communication Manager routes the call using the Call Type Digit Analysis table, taking the location of the caller into account. This field also applies to incoming trunk calls. For example, users in an organization can use five-digit extensions to make calls to internal contacts. If a user dials 8-2249, Communication Manager
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Location-Based Call Type Analysis
maps the dialed number to 1212-848-2249 using the Call Type Digit Analysis table before routing the call.
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Chapter 42: Digit Absorption
Implements up to five-digit absorption lists. Digit Absorption is required for each local telephone company central office (CO) and for a Foreign eXchange (FX) trunk group connected to a CO. Each outgoing digit string from the server running Communication Manager to the CO is treated according to entries in the Absorption Treatment Assignment section of the screen.
Available only if Digit Absorption is administered for the trunk group.
Example command: change digit-absorption n, where n is the absorption digit.
Related links Digits on page 923 Digit Absorption List on page 906
Absorption Treatment Assignment
Valid Entry A to F
Usage
The chosen treatment letter. All choices for the digits 0 through 9 must be taken from the same group (Group I or Group II).
Absorption Treatment Information
Shows how Digit Absorption treats each digit, 0 through 9, depending on the assignment of A through C for Group I, and A, D, E, and F for Group II.
List Number
Valid Entry 0 to 4
Usage The Digit Absorption List number that is referenced from the associated trunk group.
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Chapter 43: Display Parameters
Establishes how extensions of 6 to 13 digits are punctuated. Example command: change display-parameters
Related links Extension display format on page 217
EXTENSION DISPLAY FORMATS
The fields in this section of the screen override similar fields on the Dial Plan Parameters screen. If you leave these fields blank, the values on the Dial Plan Parameters screen apply.
Default Call Appearance Display Format
Affects call appearances only on telephones that support downloadable call appearance buttons, such as the 2420 and 4620 telephones. Bridged call appearances remain unaffected.
Valid Entry inter-location
intra-location
Usage
The complete extension on downloadable call appearance buttons. This is the default.
A shortened version of the extension on downloadable call appearance buttons.
Display between Locations with these Display Parameters
The system displays this field only when you enable the Multiple Locations field. This field cannot be left blank.
Valid Entry inter-location
Usage
Communication Manager uses the inter-location display format for calls between designated locations. This is the default value.
Table continues...
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Display Parameters
Valid Entry intra-location
Usage
Communication Manager uses the intra-location display format for calls within a location. You can use the intra-location format within a branch to display short extensions within that location. This Display Parameter lets you display short extensions between two or more closely associated locations. For example, if you assign display parameter 3 to locations 7 and 17, and set the field to intra-location, Communication Manager uses intra-location format for calls between locations 7 and 17, however uses inter-location calls from 7 to any other location other than 17, and for calls from 17 to any other location other than 7.
Extension Display Format
Use this field to specify how the system must punctuate extensions for inter-location and intralocation displays. Blank spaces are sometimes used in telephone extensions, especially outside of the U.S. Dots (.) are used on SAT screens in place of blanks.
The number of punctuation marks that you can use is determined by the number of xs in the format:
· If the format contains fewer than six xs, there can be no punctuation marks.
· If the format contains six or more xs, the maximum number of punctuation marks is determined by the following table.
Extension length 6 7 8 9 10 11 12 13 14 15 16
Maximum punctuation marks 2 1 3 3 3 2 1 0 0 0 0
Maximum total length 8 8 11 12 13 13 13 13 14 15 16
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Valid entry xx.xx.xx blank
EXTENSION DISPLAY FORMATS
Usage For six or more xs, characters can contain: · All x characters without punctuation · A combination of x characters and up to two hyphens · Spaces · Periods
Inter-Location
Specifies punctuation for calls between locations. This is the default.
Intra-Location
Specifies punctuation for calls within a location.
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Chapter 44: DS1 Circuit Pack
Administers all DS1 circuit packs. Example command: add ds1 n, where n is the board location.
DS1 Circuit Pack: page 1
Bit Rate
Note:
TN464C and later release circuit packs have an option switch that must be set to match this Bit Rate value.
Valid Entry 1.544 2.048
Usage The maximum transmission rate for DS1 circuit packs that support T-1 service. The maximum transmission rate for DS1 circuit packs that support E-1 service.
Channel Numbering
The ETSI and ISO QSIG specifications require that B-channels on an E1 be encoded as 1 to 30 in the Channel ID IE. Prior to the existence of this field, Communication Manager only used this scheme for Country Protocols 2a (Australia) and 13a (Germany 1TR6). Available only with ISDNPRI signaling on a private network. The interface must be peer master or peer slave.
2.048 bit rate options:
· timeslot
· sequential
If Communication Manager is connected via QSIG trunks to a switch or server supporting the ETSI QSIG or ISO QSIG specifications, this field must be sequential.
Related links Signaling Mode on page 237
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Bit Rate on page 224 Connect on page 225 Interface on page 228
DS1 Circuit Pack: page 1
Connect
To control communications at layers 2 and 3 of the ISDN-PRI protocol, this field to specifies what is on the far end of this DS1 link.
Available only for ISDN-PRI signaling.
Valid Entry pbx line-side
network
host
Usage
The DS1 link is connected to another switch in a private network.
Communication Manager is acting as the network side of an ISDN-PRI interface. Used to connect to Roll About Video equipment.
The DS1 link connects Communication Manager to a local telephone company central office or any other public network switch.
The DS1 link connects Communication Manager to a computer.
Related links Signaling Mode on page 237
Country Protocol
The country protocol used by the far-end server. For connections to a public network, your network service provider can tell you which country protocol they are using.
Available only with ISDN-PRI and CAS signaling.
Valid Entry 1 to 25
etsi
Usage
The country protocol used by the local telephone company central office at which this link terminates.
The network service provider uses the European Telecommunications Standards Institute (ETSI) protocol and the Signaling Mode is isdn-pri.
Related links Signaling Mode on page 237 Country options table on page 817
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DS1 Circuit Pack
CRC
Indicates whether a cyclic redundancy check (CRC) will be performed on transmissions that the DS1 circuit pack receives.
Valid Entry y n
Usage The Signaling Mode is CAS and the DS1 link is providing E-1 service. All other applications.
Related links Signaling Mode on page 237
D-Channel
Available only with a Japanese 2-Mbit trunk circuit pack and the ISDN-PRI Signaling Mode.
Valid Entry 1 to 31
Usage
The Japanese 2-Mbit trunk circuit pack, when administered to support ISDN-PRI signaling, assigns the D-channel to any channel from 1 to 31 in an E-1 facility.
Related links Signaling Mode on page 237
DCP/ANALOG Bearer Capability
Sets the information transfer capability in a bearer capability IE of a setup message to speech or 3.1kHz. Available only with the ISDN-PRI Signaling Mode.
Valid Entry 3.1kHz speech
Usage Provides 3.1 kHz audio encoding in the information transfer capability. Provides speech encoding in the information transfer capability.
Related links Signaling Mode on page 237
Disable Restarts
Controls whether outgoing RESTART messages are sent. Also used to disable QSIG restarts. Available when:
· Country Protocol is 3 (Japan)
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DS1 Circuit Pack: page 1
· Country Protocol is ETSI · Peer Protocol is QSIG
Valid Entry y n
Usage Outgoing restarts are disabled. In other words, RESTART messages are not sent. Outgoing RESTART messages are sent. This is the default.
DMI-BOS
The DMI/BOS protocol is used for high-speed digital communications between a host computer and Communication Manager. With this 24-channel protocol, channels 1 to 23 of the DS1 link carry data and channel 24 carries control signaling. DMI/BOS has greater capacity than a robbedbit 24-channel facility. Available only when Signaling Mode is common-chan.
Valid Entry y n
Usage Activates the Digital Multiplexed Interface-Bit Oriented Signaling (DMI-BOS) format. Uses an Avaya proprietary format.
Related links Signaling Mode on page 237
Framing Mode
Selects either superframe or extended superframe for T1 service on the DS1 link. The framing mode must match the mode used on the other end of the link. Available only with T1 service.
Tip:
Use ESF when the service provider supports it, especially if the facility gets upgraded to ISDN. The ESF format provides enhanced performance measurements and uses a sophisticated error-checking method to ensure data integrity.
Valid Entry d4 esf
Usage
The basic DS1 superframe. Use this mode only for voice traffic.
The extended superframe format. Use this mode for digital data traffic. A TN464F, TN767E, or a later suffix DS1 circuit pack requires the administration of ESF Data Link options.
Related links Bit Rate on page 224 ESF DATA LINK OPTIONS on page 241
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DS1 Circuit Pack
Idle Code
Caution:
The Country Protocol sets the default idle code. Do not change the default without assistance from Avaya or your network services provider.
Valid Entry
Usage
Any 8-digit string of Sets the signal sent out over idle DS0 channels. The string must be compatible with
0's and 1's
the protocol used by the far-end switch or server.
Related links Country Protocol on page 225
Interconnect
For E1 service using channel-associated signaling, tells Communication Manager whether the DS1 circuit pack is using a public or private network protocol. This value must agree with the Group Type value administered for the trunk group.
Available only if the Signaling Mode is CAS.
Valid Entry pbx CO
Usage
The board operates as a tie trunk circuit pack.
The board operates as a local telephone company central office (CO) or DID circuit pack. Use for Enterprise Mobility User (EMU)/EC500 administration.
Related links Signaling Mode on page 237 Group Type on page 508
Interface
Controls how the server negotiates glare with the far-end switch. The servers at either end of the DS1 link must have complementary settings in this field. Otherwise, the D-channel cannot function. For example, if the Avaya S8XXX server at one end of the link is administered as network, the other end must be administered as user. Available only when this DS1 link is providing an ISDN-PRI connection in a private network.
Related links Connect on page 225
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DS1 Circuit Pack: page 1
Private network applications in the U.S.
Valid Entry network
user
Usage
The server overrides the other end when glare occurs, and when connecting the server to a host computer.
The server releases the contested circuit and looks for another when glare occurs, and when connecting the server to a public network.
Private network applications outside the U.S.
Valid Entry peer-master peer-slave
Usage The switch overrides the other end when glare occurs. The switch releases the contested circuit and looks for another when glare occurs.
Interface Companding
The companding algorithm expected by the system at the far end.
Valid Entry a-law mu-law
Usage Algorithm expected at the far-end for E1 service. Algorithm expected at the far-end for T1 service.
Interworking Message
Determines what message Communication Manager sends when an incoming ISDN trunk call is routed over a non-ISDN trunk group.
Valid Entry PROGress
ALERTing
Usage
Requests the public network to cut through the B-channel and let the caller hear tones such as ringback or busy tone provided over the non-ISDN trunk. Normallyselected value.
Causes the public network in many countries to play ringback tone to the caller. This value is used only if the DS1 is connected to the public network, and it is determined that callers hear silence (rather than ringback or busy tone) when a call incoming over the DS1 interworks to a non-ISDN trunk.
ITN-C7 Long Timers
Controls the T302 and T303 timers. Available only if the Signaling Mode is isdn-pri.
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DS1 Circuit Pack
Valid Entry y n
Usage Increases the length of the long timers. Uses the default long timers.
Related links Signaling Mode on page 237
Line Coding
Selects the type of line coding used on this facility. The setting in this field must match the setting on the far-end of the link, or there must be an intervening CSU to convert the line coding protocols. Voice calls work even if line coding is different match, but a single data call brings down the DS1 facility.
Caution:
If you change this field, you must busy out the DS1 circuit pack. You must also change the administration for: Route-Pattern, Access Endpoint, PRI Endpoint, Signaling-Group, and Trunk-Group.
Note:
When the DS1 circuit pack is used for ISDN service, the ISDN D-channel data is inverted forami-basic or ami-zcs is used and kept upright forb8zs or hdb3 is used.
Valid Entry b8zs
ami-zcs
ami-basic hdb3 cmi
Usage
Bipolar eight zero substitution. For T1 facilities that support voice or data traffic. Provides a 64K clear channel.
Alternate mark inversion - zero code suppression. For T1 facilities that carry voice traffic. Do not use this option for digital-data applications. Use if this facility is going to be upgraded to ISDN.
Alternate mark inversion-basic. For unrestricted E1 facilities.
High density bipolar 3. For restricted E1 facilities.
Coded mark inversion. Used in Japan as the only type of line coding used with the Japanese 2 Mbit trunk circuit pack.
Line Compensation
The appropriate entry in this field varies with the type of cable used. Contact your network service provider for the correct setting.
Cable lengths for a DSX-1 cross-connect
The following valid entries are for the different lengths of 22-gauge ABAM cable terminated on a DSX-1 cross-connect.
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DS1 Circuit Pack: page 1
Valid Entry 1 2 3 4 5
Usage Length: 000 133 (ft), 000 40.5 (m) Length: 133 266 (ft), 40.5 81.0 (m) Length: 266 399 (ft), 81.0 122 (m) Length: 399 533 (ft), 122 163 (m) Length: 533 655 (ft), 163 200 (m)
Cable lengths for a DS1 interface
The following valid entries are for the different lengths of 22-gauge ABAM cable directly connected to DS1 interfaces.
Valid Entry 1 2 3 4 5
Usage Length: 0000 0266 (ft), 000 081(m) Length: 0266 0532 (ft), 081 162 (m) Length: 0532 0798 (ft), 162 243 (m) Length: 0798 1066 (ft), 243 325 (m) Length: 1066 1310 (ft), 325 400 (m)
Location
The port address.
MMI Cabling Board
Available only with the Multimedia Call Handling feature.
Valid Entry xxxxx
Usage
Location of the multimedia interface circuit pack that is connected to the Expansion Services Module (ESM). The location can be a the cabinet, carrier, or slot address.
Related links Multimedia Call Handling (Basic) on page 837 Multimedia Call Handling (Enhanced) on page 837
MMI Interface
This field displays the type of multimedia (MMI) interface. The system displays this field only when you enable the MMCH and you enter a value for the MMI Cabling Board.
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DS1 Circuit Pack
Related links Multimedia Call Handling (Basic) on page 837 Multimedia Call Handling (Enhanced) on page 837 MMI Cabling Board on page 231
Name
Assigns a significant, descriptive name to the DS1 link. Use the vendor's circuit ID for the link in this field because that information helps troubleshoot problems with the link. This field can also be used to indicate the function or the destination of this DS1 facility. Accepts up to 15 characters.
Note:
Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Peer Protocol
Administers the peer level protocol that operates in a private network. Available only if Interface is peer-master or peer-slave.
Valid Entry Q-SIG
TTC
Usage
This implements QSIG Network Basic Call. Available only if Basic Call Setup is enabled for the system.
For private networking. Requires a Digital Trunk (Japan 2 MB TTC) (TN2242) circuit pack.
Related links Basic Call Setup on page 848 Interface on page 228
Protocol Version
Available only when:
· The Signaling Mode is isdn-pri and the Connect type is network.
· The Signaling Mode is isdn-pri, the Connect typeis pbx, and the Interface type is user or network.
Valid Entry a, b, c, d
Usage
Selects the protocol that matches the network service provider's protocol in countries whose public networks allow multiple layer-3 signaling protocols for ISDNPRI service. Contact the network service provider to verify that the protocols match.
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DS1 Circuit Pack: page 1
Warning:
The AT&T Switched Network Protocol prohibits restricted displays of connected numbers. Display problems occur if you administer the 1a country-protocol/ protocol-version combination on the DS1 screen and administer the ISDN-PRI Trunk Group to restrict sending the connected number.
Related links Signaling Mode on page 237 Connect on page 225 Interface on page 228
Public network signaling administration for ISDN-PRI Layer 3
The following table describes the Communication Manager public-network access connections for ISDN-PRI Layer 3.
Admin Value 1-a
1-b
1-c 1-d 2-a
2-b 3 4 5 6 7 8 9 10-a
10-b
Country
Protocol Supported
B-channel mtce msg
United States, Canada
United States
United States United states Austrailia
Austrailia Japan Italy Netherlands Singapore Mexico Belgium Saudi Arabia United Kingdom
United Kingdom, Ireland
AT&T TR 41449/ 41459 (tested with AT&T network, Canadian network, and MCI network)
Service
Telcordia Technologies TR 1268; NIUF.302; ANSI T1.607
Restart
NORTEL DMS-250 BCS36/IEC01 Service
Telecordia Technologies SR-4287 Service
AUSTEL TS014.1; Telecom Australia TPH 1856 National ISDN protocol
Restart
ETSI ISDN protocol
Restart
NTT INS-NET
Restart
ETS 300 102
Restart
ETS 300 102
Restart
ETS 300 102
Restart
ETS 300 102
Restart
ETS 300 102
Restart
ETS 300 102
Restart
ETS 300 102 (for connection to DASS II/ DPNSS through external converter)
Restart
ETS 300 102 (Mercury); British
None
Telecom ISDN 30; Telecom Eireann
SWD 109
Table continues...
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DS1 Circuit Pack
Admin Value 11 12-a 12-b
13-a 13b 14
15 16 17 18 19 20 21 22 23 24 25 ETSI-a
ETSI-b
Country
Protocol Supported
B-channel mtce msg
Spain France France
Germany Germany Czech Republic, Slovakia Russia (CIS) Argentina Greece China Hong Kong Thailand Macedonia Poland Brazil Nordic South Africa Europe, New Zealand, etc.
Telefonica ISDN Specification VN4 (French National PRI) ETS 300 102 modified according to P10-20, called Euronumeris FTZ 1 TR 6 (German National PRI) ETS 300 102 ETS 300 102
Restart None None
None Restart Restart
ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102 ETS 300 102
Restart Restart Restart Restart Restart Restart Restart Restart Restart Restart Restart Restart
ETS 300 102
None
Received Digital Metering Pulse Maximum (ms)
Available only when the Signal Mode is cas (Channel Associated Signaling), the Interconnect type is co or pbx, and the Country Protocol is administered for a protocol that uses periodic pulse metering (PPM).
Valid Entry
20 to 1000 ms in increments of 10ms.
Usage
This value must be greater than the Received Digital Metering Pulse Minimum value and match the value used by the network services provider.
Related links Signaling Mode on page 237 Country Protocol on page 225 Received Digital Metering Pulse Minimum (ms) on page 235
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Interconnect on page 228
DS1 Circuit Pack: page 1
Received Digital Metering Pulse Minimum (ms)
Available only when the Signal Mode is cas (Channel Associated Signaling), the Interconnect type is co or pbx, and the Country Protocol is administered for a protocol that uses periodic pulse metering (PPM).
Valid Entry
Usage
20 to 1000 ms in This value must be greater than the Received Digital Metering Pulse Maximum increments of 10ms value and match the value used by the network services provider.
Related links Received Digital Metering Pulse Maximum (ms) on page 234 Signaling Mode on page 237 Country Protocol on page 225 Interconnect on page 228
Received Digital Metering Pulse Value
Available only when the Signal Mode is cas (Channel Associated Signaling), the Country Protocol is 21, and the Interconnect type is co or pbx.
Valid Entry 0, 1
Usage This value must match the value used by the network services provider.
Related links Signaling Mode on page 237 Country Protocol on page 225 Interconnect on page 228
Incoming digital PPM
The following table provides the incoming digital PPM signaling default per country protocol code.
Code 0 1 2 3 4
Country null U.S. Australia Japan Italy
PPM Min (ms) NA NA 80 NA 120
PPM Max (ms) NA NA 180 NA 150
PPM Value NA NA 0 NA 1
Table continues...
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DS1 Circuit Pack
Code 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21
22 23 24 25
Country Netherlands Singapore Mexico Belgium Saudi Arabia UK Spain France Germany Czech Republic Russia CIS Argentina Greece China Hong Kong Thailand Macedonia Croatia
Poland Brazil Nordic South Africa
PPM Min (ms) 90 NA 20 20 NA NA 20 NA NA 20 NA 10 100 NA NA 20 120
20 100 NA NA 160
PPM Max (ms) 160 NA 180 180 NA NA 220 NA NA 420 NA 180 180 NA NA 180 180
80 150 NA NA 240
PPM Value 0 NA 1 1 NA NA 0 NA NA 1 NA 1 1 NA NA 1 1
1 0 NA NA 0, 1
Side
Use this field to control how Communication Manager must resolve glare at layer 3 over an ISDNPRI link in QSIG private networks. The system displays this field only if the Interface type is peermaster or peer-slave.
Caution:
You must correctly pair with the administration of the far-end server. If the far-end server is administered as the b side, you must set the Side field to a, regardless of whether the layer 2 is peer-master or peer-slave, and vice versa.
Valid entry a
Usage
The Interface is peer-master. Communication Manager overrides the far-end when glare occurs.
Table continues...
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DS1 Circuit Pack: page 1
Valid entry b
Usage
The Interface is peer-slave. Communication Manager releases the contested circuit and looks for another when glare occurs.
Related links Interface on page 228
Signaling Mode
Selects the signaling method used for the DS1 link. This mode must match the method used by the network services provider.
Valid Entry CAS robbed-bit isdn-pri isdn-ext
common-chan
Usage
Channel Associated Signaling. Out-of band signaling with E1 service. This setting yields 30 64-kbps B-channels for voice or data transmission. Channel 0 is used for framing while channel 16 carries signaling. Used for Enterprise Mobility User (EMU)/ EC500 administration.
In-band signaling with T1 service. This setting yields 24 56-kbps B-channels for voice transmission.
Either T1 or E1 ISDN service. This setting supports both Facility Associated Signaling and Non-Facility Associated Signaling.
Either T1 or E1 ISDN service. This setting supports only Non-Facility Associated Signaling.
Note:
NFAS is primarily a feature for ISDN-T1 connections offered by service providers in North America and Hong Kong. However, it can also be used on private-network connections, and in that context it is possible to set up NFAS using ISDN-E1 interfaces.
Out-of-band signaling with T1 service. This setting yields 23 64-kbps B-channels for voice or data transmission. Channel 24 is used for signaling.
T303 Timer (sec)
Available only if the Group Type is isdn-pri.
Valid Entry 2 to 10
Usage
The number of seconds the system waits for a response from the far end before invoking Look Ahead Routing.
Related links Interface on page 228 Group Type on page 508
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DS1 Circuit Pack
MAINTENANCE PARAMETERS
Alarm When PRI Endpoint Detached
Enables or disables an alarm when the DS1 board detects a loss of signal. Used for DS1 circuit packs connected to Roll-About Video equipment.
Available only when the Connect type is line-side.
Related links Connect on page 225
Block Progress Indicator
Blocks sending the progress indicator in the SETUP message. Available only if the Country Protocol is set to 1 and the Protocol Version is set to b.
Valid Entry y n
Usage Prevents the progress indicator from being sent in the SETUP message. Supports sending the progress indicator.
Related links Country Protocol on page 225 Protocol Version on page 232
EC Configuration
The set of parameters used when cancelling echo. This information is stored in firmware on the UDS1 circuit pack.
Available only if echo cancellation is enabled for the DS1 circuit pack.
Valid Entry 1 5 to 15
2
3
Usage
Provides the most rapid adaptation in detecting and correcting echo at the beginning of a call, regardless of the loudness of the talker's voice. For very loud talkers and severe echo, the far-end talker's speech is heard as clipped when both parties talk at the same time.
Provides slightly slower adaptation to echo. Use if speech is often clipped when both parties talk at the same time.
Provides slightly slower adaptation to echo but might result in a 2 or 3 second fade on strong echo for quiet talkers. Completely removes speech clipping.
Table continues...
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DS1 Circuit Pack: page 1
Valid Entry 4
Usage
Used in cases of extreme echo, excessive clipping or breakup of speech. May result in slight echo or background noise.
Note:
For the MM710, the values 1 and 4 are reversed. That is, 1 for the MM710 is the same as 4 for the TN464HP/ TN2464CP, and 4 for the MM710 is the same as 1 for the TN464HP/ TN2464CP.
Related links Echo Cancellation on page 239
EC Direction
This field indicates the direction of the echo that is cancelled. The system displays this field only if Echo Cancellation is enabled for the DS1 circuit pack.
Valid entry inward
outward
Usage
Cancels echo energy coming back into Communication Manager. Energy from an outgoing call is reflected from an external reflection point. The party inside Communication Manager hears the echo.
Cancels echo energy going outside Communication Manager. Energy from an incoming call is reflected from an internal reflection point. The party outside Communication Manager hears the echo.
Related links Echo Cancellation on page 239
Echo Cancellation
Enables or disables echo cancellation on the Universal DS-1 circuit pack. Available only if DS1 echo cancellation is enabled on the system. Related links DS1 Echo Cancellation on page 832
Near-end CSU
Available only when the DS1 circuit pack is a TN767D or TN464E or later suffix model, with a Bit Rate of 1.544 and a Country Protocol of 1 (U.S.).
Valid Entry other integrated
Usage
No channel service unit is attached to the DS1 facility or the CSU is an external unit.
A 120A CSU module is attached to the DS1 board. This integrated channel service unit (ICSU) can accept software-administrable option downlinks. In other words, the ICSU can respond to test codes from technician's equipment and report its status. Requires administration of Integrated CSU Options.
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DS1 Circuit Pack
Related links Bit Rate on page 224 Country Protocol on page 225 INTEGRATED CSU OPTIONS on page 242
Slip Detection
Slips are synchronization errors that slow digital transmissions and can cause data loss. The server maintains a slip-count record for each DS1 interface to detect errors and evaluate their severity (the type of alarm). If as many as 50 percent of those spans administered for slip detection are experiencing slips (with respect to the primary), then a decision is made to switch to the secondary.
Caution:
Always enable slip detection for DS1 circuit packs that serve as primary or secondary synchronization references.
Valid Entry y
n
Usage
Maintenance software measures the slip-rate of this circuit pack and determines if it is excessive. Typically, for DS1 spans used for data applications and for spans used as synchronization references. This excludes all T1-spans connecting channel banks, unless the channel bank is externally timed. This entry enables switching between the primary and secondary synchronization references and an internal high-accuracy clock.
For DMI-BOS links or when testing is optional. Typically used for DS1 spans that are used exclusively for voice and that do not serve as the primary or secondary synchronization source.
DS1 circuit pack: page 2
Caution: Do not change fields on this page without assistance from Avaya or from the network service provider. For those circuit packs that support it, this page is available only if the Framing Mode is esf or the Near-end CSU type is integrated. Related links Framing Mode on page 227 Near-end CSU on page 239
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DS1 circuit pack: page 2
CPE LOOPBACK JACK OPTIONS
Supply CPE Loopback Jack Power
Enables or disables the DS1 board's ability to supply power to the equipment during loopback testing if a Customer Premise Equipment (CPE) Loopback Jack is installed.
ESF DATA LINK OPTIONS
Far-end CSU Address
Available only if the Framing Mode is esf.
Valid Entry a, b
Usage
Administers the transmit direction address used for the ESF data link command with both integrated and external channel service units (CSU).
Related links Framing Mode on page 227
Network Management Protocol
Available only if the Framing Mode is esf.
Valid Entry tabs
Usage
The data link can be remotely examined. Used only with circuit packs that have an integrated channel service unit (CSU).
Related links Framing Mode on page 227
Send ANSI-T1.403 One-Second Performance Reports
Enables or disables sending error reports from the DS1 circuit pack to the far-end server or switch. These reports are useful for network management, and are sent at 1-second intervals when enabled.
Available only if the Framing Mode is esf. It is used only with circuit packs that have an integrated channel service unit (CSU).
Go to the Avaya Support website at http://support.avaya.comhttp://support.avaya.com/ for current documentation, product notices, knowledge articles on how to use these reports.
Related links Framing Mode on page 227
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DS1 Circuit Pack
INTEGRATED CSU OPTIONS
Receive ALBO (Receive Automatic Line Build-Out)
Increases the strength of incoming signals by a fixed amount to compensate for line losses. To set correctly, measure the signal loss on this specific facility.
Valid Entry 26db 36db
Usage
Used for most applications.
Used for networks that differ from public telephone network standards, such as campus networks.
Transmit LBO (Transmit Line Build-Out)
Reduces the outgoing signal strength by a fixed amount. The appropriate level of loss depends on the distance between your Communication Manager server (measured by cable length from the smart jack) and the nearest repeater. Where another server/switch is at the end of the circuit, as in campus environments, use the cable length between the two switches to select the appropriate setting from the table below.
Available only with an integrated near-end CSU.
Valid Entry 0db -7.5db -15db -22.5db
Usage For distances of 2,001 3,000 feet For distances of 1,001 2,000 For distances of 0 1,000 feet For mid-span repeaters
Related links Near-end CSU on page 239
Upon DTE LOS
Tells Communication Manager what to do if the outgoing signal from the DS1 circuit pack, or Data Terminal Equipment (DTE), to the network is lost.
Valid Entry loopback
ais
Usage
Returns the network signal to the network. This prevents any alarms from being generated at the far-end.
Alarm Indicator Signal. Sends an unframed all-ones signal (the AIS or Blue Alarm) to the far-end server or switch. This option alerts the network service provider to the problem immediately and aids in troubleshooting.
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Chapter 45: Duplicate Station
Adds telephones by copying the information from an existing telephone and modifying this information for each new telephone. For example, configure one telephone as a template for an entire work group, and then duplicate the template station to add all the other extensions in the group. Only telephones of the same model can be duplicated. All the feature settings from the template telephone are copied to the new telephones. Example command: duplicate station n, where n is the extension number. For field descriptions, see Station on page 733.
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Chapter 46: Duplicate Vector
Duplicate vectors from an existing vector and edit the duplicate vectors to create vectors that are similar to the existing vector. You can use this functionality to configure one vector as a template that can be reused when creating similar vectors. Example command: duplicate vector n, where n is the master vector.
Assigned VDN
The first assigned VDN if a VDN was assigned to the master vector.
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More VDN's
More VDN's
Displays * if there is more than one VDN assigned to the same vector. For example, if 5555 displays in VDN Assigned and * displays in More?, this means that the master vector selected is already assigned to VDN 5555 as well as to other VDNs.
Name
The first row displays the vector name for the master vector if assigned. The following lines define the vector names for the duplications. The use of vector names is optional.
Vector
The first row displays the vector number for the master vector. The following lines define the vector numbers for the duplications. Any unassigned vector number is valid.
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Chapter 47: Enable File Transfer
Enables SFTP on TN799BP (C-LAN) and VAL circuit packs. Example command: enable filexfer
Login
The login ID. Accepts from three to six alphanumeric characters.
Password
Seven to 11 characters used as a password. The password must contain at least one number.
Secure
Enables or disables SFTP instead of FTP or TFTP. If the circuit pack does not support a secure session, no session is enabled.
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Chapter 48: Enable Session
Enables Secure SHell (SSH) instead of Telnet. Example command: enable session
Login
The login ID. Accepts from three to six alphanumeric characters.
Password
Seven to 11 characters used as a password. The password must contain at least one number.
Secure
Enables or disables SSH instead of Telnet.
Time to Login
Available only if the board is a TN2302.
Valid Entry 0 to 255
Usage The number of minutes allowed for login before the session times out.
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Chapter 49: Extended Pick-Up Group
Organizes pickup groups into extended pickup groups. The extended group is a collection of pickup groups that can answer calls from other pickup groups in the same extended group. With Extended Pick-Up Group, users can answer calls outside their immediate group. The maximum number of groups that can be added to an extended pickup group is 25. Example command: change extended-pickup-group n, where n is the group number.
Extended Group Number
The number associated with a collection of pickup groups.
Pickup Group Number
Valid Entry 1 to 5000 blank
Usage
The number for call pickup groups that can answer calls in the extended pickup group.
Pickup Number
The pickup number assigned to the pickup group. Users dial the pickup number after the feature access code (FAC) to pick up calls in their extended pickup group.
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Chapter 50: Extensions administered to have an MCT-Control button
Lists the extensions that can take control of a Malicious Call Trace (MCT) request. To give a user the ability to take control of such requests, you must add their extension to this list and assign them a mct-control feature button. Example command: display mct-group-extensions
1 to 100
The extension of a telephone or attendant console that can take control of a Malicious Call Trace (MCT). A mct-control button must be assigned to the extension station or attendant console.
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Chapter 51: Extensions to Call Which Activate Features by Name
Assigns a dialed extension to a feature within Communication Manager. This extension is called a feature name extension (FNE). The FNE mapping must be administered and all extensions must fit the dial plan. When a user calls the extension, Communication Manager activates the feature. For more information on the extensions that are paired with Feature Access Codes (FACs), see Chapter 51. For more information on the extensions that are not paired with Feature Access Codes (FACs), see Avaya Extension to Cellular User Guide. The Transfer to Voice Mail FNE is used when a user is active on a call and wants to transfer the other party to voice mail, or to the principal's voice mail, if this is a covered call. This FNE can also be used when a user goes off-hook for the first time and dials the Transfer to Voice Mail FNE to be connected to the voice mail administered in his coverage path. This is identical to dialing a Transfer to Voice Mail feature access code (FAC). Example command: change off-pbx-telephone feature-name-extensions
Extension
Any valid and assigned extension number. Users call it from Cellular telephones to access Communication Manager features. Dialing the extension from their Cellular telephone is equivalent to dialing the FAC administered for the associated feature, or activating the associated feature.
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Chapter 52: Failover Group Domain Mapping
A Failover Group is a group of two active Session Manager instances that are interconnected to ensure high availability of Session Manager services. Route pattern and Failover Group domain mapping are used for call preservation during a network outage. You can enter up to 9 domains in the Failover Group Domain Mapping table. For more information on call preservation administration, see Call Preservation Feature Description and Administration Guide. Example command: display failover-grp-domain-map
Domain
Valid Entry A fully qualified domain name
blank
Usage
You must type the same primary and the inverse failover group domain names, which you have already set on the Session Manager Failover Group screen. For example, if you have set the Failover Group Domain names on the Session Manager Failover Group screen as SM1-SM2.company.com (primary domain name) and SM1-SM2-inverse. company.com (inverse domain name), must set the Failover Group Domain names on the Communication Manager Failover Group screen as the same.
The maximum length of the field is 64 characters.
Note:
If you enter a domain name, specify Route Pattern. If you set the Domain field to blank, set Route Pattern to blank.
This is the default.
Related links Route Pattern on page 252
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Failover Group Domain Mapping
Route Pattern
Valid Entry 12000
blank
Usage
If you are using the Avaya supported server, you can enter 1999 possible route patterns. You must set the same route pattern in the order you defined on the Session Manager Failover Group screen. For example, if you have set the SM2 as first Session Manager and SM1 as secondary Session Manager in the route pattern list, on the Communication Manager Failover Group screen, the route pattern must have the trunk group that is associated to SM2 listed first and the trunk for SM1 listed second.
You must set the Group Type field on the Trunk Group and the Signaling Group screens to sip and set the Peer Server field on the Signaling Group screen to SM for the referenced route number.
For the Avaya supported servers, see Avaya Aura® Communication Manager Hardware Description and Reference.
Note:
If you enter Route Pattern, do not set the Domain field to blank.
This is the default.
Related links Domain on page 251 Group Type on page 708 Group Type on page 508
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Chapter 53: Feature Access Code (FAC)
Assigns feature access codes (FACs) that, when dialed, activate or cancel system features. Example command: change feature-access-codes
Feature Access Code: page 1
Abbreviated Dialing List1 Access Code
A feature access code (FAC) used to access AD list 1. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Abbreviated Dialing List2 Access Code
A feature access code (FAC) used to access AD list 2. This value must conform to the FACs or dial access codes defined by the dial plan.
Abbreviated Dialing List3 Access Code
A feature access code (FAC) used to access AD list 3. This value must conform to the FACs or dial access codes defined by the dial plan.
Abbreviated Dial - Prgm Group List Access Code
FAC used to enter a group list from a telephone. The user's extension must be administered with permission to program the group list. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Program Ext on page 30
Announcement Access Code
FAC used to record announcements. This value must conform to the FACs or dial access codes defined by the dial plan.
Answer Back Access Code
FAC used to retrieve parked calls. If no one answers the call before a system-wide expiration interval expires, the system redirects the call. This value must conform to the FACs or dial access codes defined by the dial plan.
Auto Route Selection (ARS) Access Code 2
Additional FAC used to access ARS. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code: page 1
Attendant Access Code
FAC used to call the attendant. While only one attendant can be administered for the dial plan, more than one attendant FAC can be administered in a single distributed network. Attendant access numbers can start with any number from 0 to 9 and contain one or two digits. Available only if an attendant call type is not administered for the dial plan. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Call Type on page 210
Auto Alternate Routing (AAR) Access Code
FAC used to access AAR. AAR routes calls to a different route than the first-choice route when facilities are unavailable. This value must conform to the FACs or dial access codes defined by the dial plan.
Automatic Callback Activation/Deactivation
FAC used to activate or cancel Automatic Callback. Automatic Callback enables internal callers, upon reaching a busy extension, to have the system automatically connect and ring both originating and receiving parties when the receiving party becomes available. This value must conform to the FACs or dial access codes defined by the dial plan.
Auto Route Selection (ARS) Access Code 1
FAC used to access ARS. The system can automatically choose the least-expensive way to send a toll call. You can have one ARS access code for local and one for long distance, and route accordingly. This value must conform to the FACs or dial access codes defined by the dial plan.
Call Forwarding Activation Busy/DA
FAC used to forward calls to an administered number if the user is busy or does not answer. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Call Forwarding Enhanced Activation/Deactivation
FAC numbers used to activate and deactivate Enhanced Call Forwarding. Enhanced Call Forwarding forwards incoming calls to different destinations depending on whether they are from internal or external sources. The FACs for activation and deactivation must be administered together. This value must conform to the FACs or dial access codes defined by the dial plan.
Call Forwarding Enhanced Status
FAC used to display the status of Enhanced Call Forwarding. This value must conform to the FACs or dial access codes defined by the dial plan.
Call Park Access Code
FAC used to park an active call that can then be retrieved from a different station using the answer back access code. The call park access code cannot have the same first digit as another feature access code that is longer in length. This value must conform to the FACs or dial access codes defined by the dial plan.
Call Pickup Access Code
FAC used to answer a call directed to a pickup group. This value must conform to the FACs or dial access codes defined by the dial plan.
CAS Remote Hold/Answer Hold-Unhold Access Code
FAC used by a Centralized Attendant Service (CAS) attendant to place calls on hold and answer calls held at a remote server running Communication Manager. This FAC can also be used by an analog station. Flashing the switch-hook for the proper interval (between 200 and 1000 ms) while talking on an existing call causes the existing call to be placed on soft hold, using which the analog user can dial the Answer Hold-Unhold FAC to Hard hold the call. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code: page 1
CDR Account Code
FAC used prior to entering an account code for Call Detail Recording (CDR) purposes. CDR is a feature that uses software and hardware to record call data. This value must conform to the FACs or dial access codes defined by the dial plan.
Change COR Access Code
Using FAC, users can change their class of restriction (COR) from a telephone. Available only if Change COR by FAC is enabled for the system. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Change COR by FAC on page 831
Change Coverage Access Code
FAC used to change a coverage path from a telephone or remote station. The coverage path is the order in which calls are redirected to alternate answering positions. An extension must have station Security Codes administered to use this FAC. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Security Code on page 753
Contact Closure Close Code
FAC used to close a contact closure relay. Contact closures control electrical devices remotely. Users use an FAC to activate electrical devices such as electrical door locks. If Contact Closure Open Code is administered, then Contact Closure Close Code must also be administered. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Contact Closure Open Code on page 258
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Feature Access Code (FAC)
Contact Closure Open Code
FAC used to open a contact closure relay. Contact closures control electrical devices remotely. Users use an FAC to activate electrical devices such as electrical door locks. If Contact Closure Close Code is administered, then Contact Closure Open Code must also be administered. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Contact Closure Close Code on page 257
Feature Access Code: page 2
Contact Closure Pulse Code
FAC used to pulse a contact closure relay. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code: page 2
Data Origination Access Code
FAC used to originate a data call from a voice station. This value must conform to the FACs or dial access codes defined by the dial plan.
Data Privacy Access Code
FAC used to isolate a data call from call waiting or other interruptions. This value must conform to the FACs or dial access codes defined by the dial plan.
Directed Call Pickup Access Code
The FAC used with the extension number for which the user wants to pick up a call. To pick up a call for an extension, the user dials the FAC followed by the extension number. The called extension number stops ringing and the user connects to the call for pickup. The FAC value must conform to the FACs or dial access codes defined in the dial plan.
Directed Group Call Pickup Access Code
The FAC used to pickup a call from any pickup group. The FAC value must conform to the FACs or dial access codes defined in the dial plan.
EC500 Self Administration Access Code
Using FAC, users can self-administer their cell phone number for the Extension to Cellular feature. Users can add or change their cell phone number through this feature access code. An administrator can still enter or change cell phone numbers. The user calls the Self Administration Access Code access code and enters their cell phone number. This value must conform to the FACs or dial access codes defined by the dial plan.
Emergency Access To Attendant Access Code
FAC used to gain access to the attendant in an emergency. Such calls alert as emergency calls. Available only if Emergency Access to Attendant is enabled for the system. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Related links Emergency Access to Attendant on page 833
Enhanced EC500 Activation
Using FAC, users can activate Extension to Cellular remotely. This value must conform to the FACs or dial access codes defined by the dial plan.
Enhanced EC500 Deactivation
Using FAC, users can deactivate Extension to Cellular remotely. This value must conform to the FACs or dial access codes defined by the dial plan.
Enterprise Mobility User Activation
FAC used to activate the Enterprise Mobility User feature for a particular user, associating the features and permissions of their primary telephone to a telephone of the same type anywhere within the customer's enterprise. This value must conform to the FACs or dial access codes defined by the dial plan.
Enterprise Mobility User Deactivation
FAC used to deactivate the Enterprise Mobility User feature. This value must conform to the FACs or dial access codes defined by the dial plan.
Extended Call Fwd Activate All
FAC used to activate call forwarding from a telephone or remote location. An extension must have station Security Codes administered to use this FAC. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Security Code on page 753
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Feature Access Code: page 2
Extended Call Fwd Activate Busy D/A
FAC used to activate call forwarding from a telephone or remote location. An extension must have station Security Codes administered to use this FAC. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Security Code on page 753
Extended Call Fwd Deactivation
FAC used to deactivate call forwarding from a telephone or remote location. An extension must have station Security Codes administered to use this FAC. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Security Code on page 753
Extended Group Call Pickup Access Code
FAC used to answer a call directed to another pickup group. Users must enter a valid pickup number following this field to complete the operation. This value must conform to the FACs or dial access codes defined by the dial plan.
Facility Test Calls Access Code
FAC used to place a facility test call. This value must conform to the FACs or dial access codes defined by the dial plan.
Security alert: To ensure the security of your system, leave this field blank except when actually testing trunks.
Flash Access Code
FAC used to generate trunk flash. This code ensures that the flash signal is interpreted by the local telephone company central office switch, rather than by Communication Manager.
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Feature Access Code (FAC)
This value must conform to the FACs or dial access codes defined by the dial plan.
Group Control Restrict Activation/Deactivation
FAC used to change the restriction level for all users with a given class of restriction. Requires console permissions. This value must conform to the FACs or dial access codes defined by the dial plan.
Hunt Group Busy Activation/Deactivation
Hunt group members use the Hunt Group Busy Activation FAC to make the extension unavailable and the Hunt Group Busy Deactivation FAC to make the extension available. The hunt group member must dial:
· a two-digit hunt group number for a hunt group supporting up to 99 extensions (small hunt group) or
· a four-digit hunt group number for a hunt group supporting up to 8000 extensions (large hunt group)
If needed, the hunt group member must add zeros before the hunt group number to ensure that the small hunt group has a two-digit number and the large hunt group has a four-digit number. This value must conform to the FACs or dial access codes defined by the dial plan.
ISDN Access Code
FAC used to place an ISDN call without using ARS, AAR, or UDP.
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Feature Access Code: page 2
This value must conform to the FACs or dial access codes defined by the dial plan.
Last Number Dialed Access Code
FAC used to redial the last number dialed from this station. This value must conform to the FACs or dial access codes defined by the dial plan.
Leave Word Calling Message Retrieval Lock
FAC used to lock the display module on telephones. The lock function activates at a telephone by dialing this system-wide lock access code. This prevents unauthorized users from displaying, canceling, or deleting messages associated with the telephone. Available only if Lock Messages is administered for the station. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Lock Messages on page 746
Leave Word Calling Message Retrieval Unlock
FAC used to unlock a telephones display module. The lock function is canceled at the telephone by dialing this unlock FAC followed by the SCC. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Feature Access Code: page 3
Leave Word Calling Cancel A Message
FAC used to cancel a leave word calling message. This value must conform to the FACs or dial access codes defined by the dial plan.
Leave Word Calling Send A Message
Using FAC, users can send a leave word calling message that allows internal system users to leave a short pre-programmed message for other internal users. This value must conform to the FACs or dial access codes defined by the dial plan.
Limit Number of Concurrent Calls Activation/Deactivation
Use this field to configure the feature access codes to enable or disable the Limit Number of Concurrent Calls feature. If a user dials the activation FAC, Communication Manager restricts the number of incoming calls on the station to one call at a time. If the station user is busy, the subsequent incoming calls receive a busy tone or are transferred to voice mail.
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Feature Access Code: page 3
Note: Communication Manager Release 6.2 and earlier supported the Limit Number of Concurrent Calls feature on H.323 and DCP stations. Communication Manager Release 6.3.6 extends this support to SIP stations.
Malicious Call Trace Activation
FAC used to activate a trace request on a malicious call. This value must conform to the FACs or dial access codes defined by the dial plan.
Meet-me conference access code change
Using FAC, the controlling user of a Meet-me Conference VDN can change the access code. The Meet-me Conference feature is used to set up a dial-in conference of up to 6 parties.
Note: For 12 parties to participate in a conference, you must enable the 12party Conferences field in the Feature-Related System-Parameters screen. This value must conform to the FACs or dial access codes defined by the dial plan.
Message Sequence Trace (MST) Disable
Provides the ability to disable the MST traces using a Feature Access Code.
PASTE (Display PBX data on telephone) Access Code
FAC used to view call center data on display telephones. PASTE is used in conjunction with Avaya IP Agent. This value must conform to the FACs or dial access codes defined by the dial plan.
Per Call CPN Blocking Code Access Code
Use this FAC to activate Calling Party Number (CPN) blocking of a station. When users dial the code, the calling party information is not sent to the public network. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Per Call CPN Unblocking Code Access Code
Use this FAC to deactivate Calling Party Number (CPN) blocking of a station. When users dial the code, the calling party information is sent to the public network. This value must conform to the FACs or dial access codes defined by the dial plan.
Personal Station Access (PSA) Associate Code
FAC used to associate a telephone with the telephone features assigned to a users extension. Available only if Personal Station Access (PSA) is enabled for the system. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Personal Station Access (PSA) on page 838
Personal Station Access (PSA) Dissociate Code
FAC used to remove the association between a physical telephone and an extension number. Available only if Personal Station Access (PSA) is enabled for the system. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Personal Station Access (PSA) on page 838
Posted Messages
FAC used to access the Posted Messages feature. The Posted Messages feature provide callers with a displayed message on the telephone that states why the user is unavailable to take a call. Available only if the Posted Messages feature is enabled for the system. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Posted Messages on page 839
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Feature Access Code: page 3
Priority Calling Access Code
FAC used to enable priority calling, a special type of call alerting between internal telephone users, including the attendant. The called party hears a distinctive ringing when the calling party uses Priority Calling. This value must conform to the FACs or dial access codes defined by the dial plan.
Program Access Code
FAC used to program abbreviated dial buttons on an individual telephone. This value must conform to the FACs or dial access codes defined by the dial plan.
Refresh Terminal Parameters Access Code
Feature Access Code (FAC) is used to request a refresh of the terminal parameters on a telephone that supports downloadable parameters. This FAC is used after a DCP telephone is installed or replaced to ensure that all the terminal parameters, including button labels, are sent to the telephone.
Remote Send All Calls Activation/Deactivation
FAC used to activate or deactivate the Send All Calls feature. Requires console permissions. This value must conform to the FACs or dial access codes defined by the dial plan.
Self Station Display Activation
FAC used on a digital station to display its primary extension number when the FAC is entered. This value must conform to the FACs or dial access codes defined by the dial plan.
Send All Calls Activation/Deactivation
FAC used to activate or deactivate sending all calls to coverage with minimal or no alerting at the station. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Station Firmware Download Access Code
FAC used for 2420/2410 DCP station firmware downloads. This value must conform to the FACs or dial access codes defined by the dial plan.
Feature Access Code: page 4
3PCC H323 Override SIP Station Activation/Deactivation
FAC used to activate and deactivate third party call control (3PCC) redirect actions from a SIP desktop station to a remote softphone or virtual private network H.323 telephone. This value must conform to the FACs or dial access codes defined by the dial plan.
PIN Checking for Private Calls Access Code
This FAC enables the PIN Checking for Private Calls feature. Use this feature to allow users to make calls that are restricted and that prompt users to enter a Personal Identification Number
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Feature Access Code: page 4
(PIN) after dialing the FAC. The system displays this field only if PIN Checking for Private Calls is enabled. This value must conform to the FACs or dial access codes defined by the dial plan. Related links PIN Checking for Private Calls on page 364
PIN Checking for Private Calls Using AAR Access Code
This FAC enables the PIN Checking for Private Calls feature. Use this feature to allow users to make calls that are restricted and prompt users to enter a Personal Identification Number (PIN) after dialing the FAC. The system displays this field only if PIN Checking for Private Calls is enabled. This value must conform to the FACs or dial access codes defined by the dial plan. If a user dials and assigns this FAC, the Call Processing feature treats the call as if the user has dialed the FAC for AAR explicitly after entering the PIN code. Related links PIN Checking for Private Calls on page 364
PIN Checking for Private Calls Using ARS Access Code
This FAC enables the PIN Checking for Private Calls feature. Use this feature to allow users to make calls that are restricted and prompt users to enter a Personal Identification Number (PIN) after dialing the FAC. The system displays this field only if PIN Checking for Private Calls is enabled. This value must conform to the FACs or dial access codes defined by the dial plan. If a user dials and assigns this FAC, the Call Processing feature treats this call as if the user has dialed the FAC for ARS explicitly after entering the PIN code. Related links PIN Checking for Private Calls on page 364
Station Lock Activation/Deactivation
FAC used to activate or deactivate Station Lock. The Station Lock feature locks a telephone to prevent others from placing outgoing calls from the telephone. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Station Security Code Change Access Code
FAC used to change a Station Security Code (SSC). The SSC feature is used to deny other users access to the functions that are associated with the station. Each station user can change their own SSC if they know the current settings for the station. The SSC must be administered before the user can change it using this FAC. This value must conform to the FACs or dial access codes defined by the dial plan.
Station User Admin of FBI Assign
FAC used to activate or deactivate Facility Busy Indicators that provide visual indicators of the busy or idle status of any particular trunk group, hunt group member, or station user. This value must conform to the FACs or dial access codes defined by the dial plan.
Station User Button Ring Control Access Code
FAC used to control the ring behavior for each line appearance and bridged appearance from the station. Users can have their telephones ring either silently or audibly. This value must conform to the FACs or dial access codes defined by the dial plan.
Terminal Dial-Up Test Access Code
FAC used to perform tests on digital telephones to make sure that the telephone and the buttons are communicating properly with the server running Communication Manager. The Terminal DialUp test ensures that the terminal and each of its buttons can communicate with the server. This test is initiated by a user entering this feature access code. This test is mostly for use by terminal service personnel, but can be used by any station user. This value must conform to the FACs or dial access codes defined by the dial plan.
Terminal Translation Initialization Merge Code
FAC used to install, or merge, a station without losing any of its previous feature settings. The Terminal Translation Initialization Separation Code must already have been activated or the station administered without hardware, when the telephone was removed from its former location in order for the Terminal Translation Initialization Merge Code to be effective. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code: page 4
Terminal Translation Initialization Separation Code
FAC used to remove, or separate, a station from a location without losing any of its feature settings. This value must conform to the FACs or dial access codes defined by the dial plan.
Transfer to Voice Mail Access Code
FAC used to activate coverage to transfer the caller to the original call recipient's voice mail where the caller can leave a message. This FAC cannot have the same first digit as another FAC that is longer in length. This value must conform to the FACs or dial access codes defined by the dial plan.
Trunk Answer Any Station Access Code
The FAC used to answer calls alerting on night bells. This value must conform to the FACs or dial access codes defined by the dial plan.
User Control Restrict Activation/Deactivation
FAC used to activate and deactivate specific restrictions for an individual user or an attendant. Requires console permissions. This value must conform to the FACs or dial access codes defined by the dial plan.
Voice Coverage Message Retrieval Access Code
FAC used to retrieve voice messages for another user used as a coverage point, using a digital display module. This value must conform to the FACs or dial access codes defined by the dial plan.
Voice Principal Message Retrieval Access Code
FAC used by a user to retrieve their own voice messages for another user using a digital display module. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Whisper Page Activation Access Code
FAC used to place a page to another user's telephone when active on a call. Only the paged user hears the page, not the other parties on the call. This value must conform to the FACs or dial access codes defined by the dial plan.
Feature Access Code: page 5
AACC Conference Modes
Avaya Aura® Contact Center uses the Agent Mode feature access codes to restrict the call transfers or conferences that an agent can initiate. In the Restrict First Consult mode, an agent can initiate only one conference or one transfer sequence at a time. When an agent initiates a call transfer or a conference, the system matches the held call appearance and the initiated call appearance. Before initiating a new conference or a transfer sequence, the agent must either complete the transfer with one of the two appearances or drop one of the calls.
Note:
The AACC Conference modes are not supported on ISDN-BRI telephones.
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Feature Access Code: page 5
Restrict First Consult Activation/Deactivation
Use this field to enable the Restrict Call Joining feature. The Restrict Call Joining feature is supported only on Avaya Aura® Contact Center. With this feature, Communication Manager restricts the agents from initiating a transfer or a conference operation.
Restrict Second Consult Activation/Deactivation
Use this field to enable the Restrict Second Agent Consult feature. This feature is supported only on Avaya Aura® Contact Center. With this feature, Communication Manager allows the agents to use only one consult operation at a time. The consult operations include transfer and conference.
Note: If you set the Restrict Second Call Consult field to n, after you dial the Restrict Second Consult feature access code, the feature access code overrides the setting of the Restrict Second Call Consult field on the COR group of the station.
Add Agent Skill Access Code
FAC dialed by an agent to add a skill to their current skill set. This value must conform to the FACs or dial access codes defined by the dial plan.
After Call Work Access Code
FAC dialed by an agent when the agent performs work-related Automatic Call Distribution (ACD) activities. This value must conform to the FACs or dial access codes defined by the dial plan.
Assist Access Code
FAC dialed by an agent to request assistance from the split supervisor. The split supervisor is someone working in a call center that uses various Communication Manager and Call Management System features to check split and agent performance and to provide assistance if necessary. This value must conform to the FACs or dial access codes defined by the dial plan.
Auto-In Access Code
FAC dialed when an agent is ready to process another call as soon as the current call is completed.
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Feature Access Code (FAC)
This value must conform to the FACs or dial access codes defined by the dial plan.
Aux Work Access Code
FAC used when an agent is unavailable to receive ACD, or work-related, calls. Agents use this FAC for activities such as taking a break, going to lunch, or placing an outgoing call. This value must conform to the FACs or dial access codes defined by the dial plan.
Login Access Code
FAC dialed by an agent to gain access to the ACD functions. This is a system-wide code for all ACD agents. This value must conform to the FACs or dial access codes defined by the dial plan.
Logout Access Code
FAC dialed by the agent to exit ACD. This is a system-wide logout code for all ACD agents. This value must conform to the FACs or dial access codes defined by the dial plan.
Manual-In Access Code
FAC dialed when an agent is ready to process another call manually. This value must conform to the FACs or dial access codes defined by the dial plan.
Remote Logout of Agent Access Code
FAC typically used by a supervisor to logout an idle agent without being physically present at the agent station. The supervisor can be locally or remotely located. Available only if Service Observing (Remote/By FAC), Vectoring (Basic), and Vectoring (Prompting) are enabled on the system. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Service Observing (Remote/By FAC) on page 845 Vectoring (Basic) on page 845 Vectoring (Prompting) on page 846
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Feature Access Code: page 5
Remove Agent Skill Access Code
FAC dialed by an agent to remove a skill from their current skill set. Available only if Service Observing (Remote/By FAC) is enabled on the system. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Service Observing (Remote/By FAC) on page 845
Service Observing Listen Only Access Code
Dial FAC for a station with Service Observing permission to listen to other agent ACD calls without being heard on the ACD call. This value must conform to the FACs or dial access codes defined by the dial plan.
Service Observing Listen/Talk Access Code
Dial FAC for a station with Service Observing permission to both listen and be heard on an ACD call. This value must conform to the FACs or dial access codes defined by the dial plan.
Service Observing Next Call Listen Only Access Code
You can assign a Feature Access Code (FAC) to the Service Observing Next Call Listen Only Access Code feature. When you dial the Service Observing Next Call Listen Only Access Code, Communication Manager waits until the current call ends that the extension to be service observed is on. Communication Manager then starts service observing at the beginning of that station's next call. You can listen to the other call, but not speak into it.
Service Observing No Talk Access Code
Dial FAC for a station with Service Observing permission to listen only. Any attempt to toggle between listening and talking using the Service Observing button is denied. Available only if Expert Agent Selection (EAS) Enabled is administered for the system. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Expert Agent Selection (EAS) Enabled on page 330
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Feature Access Code (FAC)
Feature Access Code: page 6
Converse Data Return Code
FAC used to pass values between a Voice Response Unit (VRU) and Communication Manager to play an announcement, collect digits from the caller, and so on.
Vector Variable x
FAC (# can be used as the first digit), used to change the value of defined variables. Related links
Variables for Vectors on page 987
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Feature Access Code: page 7
Feature Access Code: page 7
Automatic Wakeup Call Access Code
FAC dialed to schedule or cancel a wakeup call. This value must conform to the FACs or dial access codes defined by the dial plan.
Housekeeping Status (Client Room) Access Code
FAC the housekeeper dials from a client room to provide room status. These codes are transmitted to the Property Management System (PMS) for processing. This value must conform to the FACs or dial access codes defined by the dial plan.
Housekeeping Status (Station) Access Code
FAC the housekeeper dials to provide room status. This access code must be dialed from designated telephones. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code (FAC)
Verify Wakeup Announcement Access Code
FAC dialed to verify a wakeup announcement. This value must conform to the FACs or dial access codes defined by the dial plan.
Voice Do Not Disturb Access Code
FAC dialed to enter or cancel a "do not disturb" request using voice prompting. This value must conform to the FACs or dial access codes defined by the dial plan.
Feature Access Code: page 8
Basic Mode Activation
FAC dialed to revert an Enhanced multimedia complex to a Basic multimedia complex. This value must conform to the FACs or dial access codes defined by the dial plan.
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Feature Access Code: page 8
Enhanced Mode Activation
FAC used to convert a Basic multimedia complex to an Enhanced multimedia complex. This value must conform to the FACs or dial access codes defined by the dial plan.
Multimedia Call Access Code
FAC that indicates to Communication Manager that an Enhanced mode multimedia call is being made. This FAC originates a multimedia call according to system-wide default settings. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Default Multimedia Outgoing Trunk Parameter Selection on page 290
Multimedia Data Conference Activation
FAC that when entered from any voice station that is participating in a multimedia call, alerts Communication Manager to enable data collaboration with the other parties on the call. If this FAC is entered a second time, multimedia data conference activation is denied since it is already active. This FAC only applies to voice stations on servers equipped with ESM adjuncts. This value must conform to the FACs or dial access codes defined by the dial plan.
Multimedia Data Conference Deactivation
FAC that when entered from the telephone deactivates the data session and reverts to a voice and video call. This telephone supports data collaboration on a multimedia mode call. If a user enters this FAC while participating in a data collaboration multimedia call that the user did not initiate, the system denies the deactivation. This value must conform to the FACs or dial access codes defined by the dial plan.
Multimedia Multi-Address Access Code
FAC that supports origination of a multimedia call from a voice station. It is used when the destination being dialed requires a different address for each of the 2B-channels. For example, ISDN-BRI provided by a local telephone company central office is provisioned with separate listed directory numbers for each B-channel. To make a 2B multimedia call to such a device, two sets of addresses must be entered.
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Feature Access Code (FAC)
Originating a multimedia call with the multimedia multi-address access code originates a call according to system-wide default settings. This value must conform to the FACs or dial access codes defined by the dial plan. Related links Default Multimedia Outgoing Trunk Parameter Selection on page 290
Multimedia Parameter Access Code
FAC that prompts Avaya Communication Manager to initiate a multimedia mode call with a specific bearer capability. This FAC would be followed by a 1 or 2 to indicate the following parameter selections respectively: 2x64 (unrestricted initial system default), 2x56 (restricted). This value must conform to the FACs or dial access codes defined by the dial plan.
Feature Access Code: page 9
Precedence Calling Access Code
FAC used to access the Multiple Level Precedence and Preemption (MLPP) feature using which users can request priority processing of calls during critical situations.
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Feature Access Code: page 9
This value must conform to the FACs or dial access codes defined by the dial plan.
WNDP PRECEDENCE ACCESS CODES
FAC used to determine the precedence level for a call when the Worldwide Numbering Dial Plan (WNDP) feature is active. The WNDP feature is compatible with the standard numbering system that the Defense Communications Agency (DCA) established. Different feature access codes are assigned for each PRECEDENCE level. This value must conform to the FACs or dial access codes defined by the dial plan.
Flash Override Access Code
FAC that corresponds to the Flash Override preemption level. This value must conform to the FACs or dial access codes defined by the dial plan. This field is visible only if Multi level precedence preemption is set to Y on the system-features customer options form.
Flash Access Code
FAC that corresponds to the Flash preemption level. This value must conform to the FACs or dial access codes defined by the dial plan.
Immediate Access Code
FAC that corresponds to the Immediate preemption level. This value must conform to the FACs or dial access codes defined by the dial plan.
Priority Access Code
FAC that corresponds to the Priority preemption level. This value must conform to the FACs or dial access codes defined by the dial plan.
Routine Access Code
FAC that corresponds to the Routine preemption level. This value must conform to the FACs or dial access codes defined by the dial plan.
Destination Code Control
Note: This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is
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Feature Access Code (FAC)
available only for the USA Department of Defense (DoD) and approved Federal government customers. Control DCC activation and deactivation by dialing the Activation code and the Deactivation code. DCC activates and deactivates the functionality systemwide.
Activation Code
Note: This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers. The FAC used to activate the DCC feature. This value must conform to the FACs defined by the dial plan.
Deactivation Code
Note: This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers. The FAC used to deactivate the DCC feature. This value must conform to the FACs defined by the dial plan.
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Chapter 54: Feature-related system parameters
This screen implements system parameters associated with various system features. Note: Call Coverage and Call Forwarding parameters are located on the System Parameters Call Coverage / Call Forwarding screen.
Feature-related system parameters: page 1
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Feature-related system parameters
AAR/ARS Dial Tone Required
Enables or disables a second dial tone that tells the user that additional dialing can occur. The second dial tone is provided on a incoming tie or DID trunk call routed through AAR/ARS.
Abbreviated Dial Programming by Assigned Lists
Valid Entry y n
Usage
Programming is done by the station-assigned list.
Indicates that a Program Access code is being used to indicate which personal list is to be programmed.
ACA Long Holding Time Originating Extension
The extension number that the ACA feature uses when sending a long holding time referral call. This extension number must be different from the extension number used for ACA Short Holding Time Originating Extension. Available for local or primary ACA Referral Calls. Available only when Automatic Circuit Assurance (ACA) Enabled is enabled for the system.
ACA Referral Calls
Indicates where Automatic Circuit Assurance (ACA) referral calls generate. The Automatic Circuit Assurance (ACA) feature is used to identify possible trunk malfunctions. If a possible trunk malfunction is identified, Communication Manager sends the attendant a referral call that consists of a display message or a voice-synthesized message indicating the problem and where the problem is located. Available only when Automatic Circuit Assurance (ACA) Enabled is enabled for the system.
Valid Entry local primary
remote
Usage
Generate on and for the local switch.
Generate on the local switch for remote servers and switches as well as the local switch.
Generate at another server in a DCS network. In this case, the remote node number must also be entered. The remote node number is the same node number administered for the dial plan. Also, ACA button status transmits to other servers and switches when in a DCS network.
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Feature-related system parameters: page 1
ACA Referral Destination
The specified extension or attendant must be equipped with a display module. Available only for local or primary ACA Referral Calls. Available only when Automatic Circuit Assurance (ACA) Enabled is enabled for the system.
Valid Entry extension number
attd
Usage
The extension on a local server running Communication Manager receives the ACA referral call.
The attendant receives the ACA referral call.
ACA Remote PBX Identification
Available for remote ACA Referral Calls. Available only when Automatic Circuit Assurance (ACA) Enabled is enabled for the system.
Valid Entry 1 to 63
Usage
Identifies the switch in a DCS network that makes the referral call. The switch identified here cannot be the remote server or switch identified as local on the system Dial Plan.
ACA Short Holding Time Originating Extension
The extension number that the ACA feature uses when sending a short holding time referral call. This extension number must be different from the extension number used for ACA Long Holding Time Originating Extension.
Available for local or primary ACA Referral Calls. Available only when Automatic Circuit Assurance (ACA) Enabled is enabled for the system.
Auto Abbreviated/Delayed Transition Interval (rings)
Valid Entry 1 to 16
Usage
The number of rings before an automatic abbreviated or delayed transition is triggered for a call.
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Feature-related system parameters
Automatic Callback -- No Answer Timeout Interval (rings)
Valid Entry 2 to 9
Usage
The number of times the callback rings at the calling station before the callback is canceled. Using Automatic Callback, internal users who place a call to a busy or an unanswered internal telephone can be called back when the called telephone becomes available.
Automatic Circuit Assurance (ACA) Enabled
Enables or disables the ACA feature to measure the holding time of each trunk call to identify possible trunk malfunctions. Long duration thresholds are designed to detect trunks that have been connected beyond the limit. Short duration thresholds detect trunks that might be dropping. Requires that an aca-halt button is administered on the user station.
Call Park Timeout Interval (minutes)
Valid Entry 1 to 90
Usage
The number of minutes a call remains parked before it cancels. Using the Call Park, users can retrieve a call on hold from any other telephone within the system.
DID/Tie/ISDN/SIP Intercept Treatment
Valid Entry announcement
attendant vdn
Usage Routes intercepted calls to a recorded announcement that indicates that the number reached is inoperative or restricted.
Note:
This intercept treatment is provided as a system option. Only one message is given to the user. Routes intercept calls to the attendant so that the attendant knows when a problem exists. Routes intercepted calls to a vector directory number.
Display Calling Number for Room to Room Caller ID Calls
Enables or disables displaying the calling number for room to room hospitality calls.
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Feature-related system parameters: page 1
Internal Auto-Answer of Attd-Extended/Transferred Calls
Using Internal Automatic Answer (IAA), users can answer internal calls on digital telephones (except BRI) using a speakerphone or a headset.
Valid entry attd-extended both none transferred
Usage Enables IAA for only attendant-extended calls. Enables IAA for station-transferred and attendant-extended calls. Disables IAA for all calls. Enables IAA for only station-transferred calls.
Music (or Silence) On Transferred Trunk Calls
Valid Entry all no
call-wait
Usage
All transferred trunk calls receive music until the call is answered if the Music-onHold feature is available.
When the Trunk callers are waiting to be answered, they hear music or nothing if Music-on-Hold is not administered. When the transfer is completed, the callers then hear ringback till the call is answered.
Trunk calls are transferred to stations that require the call to wait and hear music, if administered. All other transferred trunk calls receive ringback tone.
Music/Tone on Hold
Indicates what a caller hears while on hold. Available only if Tenant Partitioning is disabled for the system.
Valid Entry music tone none
Usage The caller hears music while on hold. The caller hears a tone while on hold. The caller hears silence while on hold.
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Feature-related system parameters
Off-Premises Tone Detect Timeout Interval (seconds)
Valid Entry 5 to 25
Usage
The number of seconds a call progress tone receiver (CPTR) tries to detect dial tone from a trunk during dialing. Once the time-out interval occurs, the call either outpulses on the trunk or gets intercept treatment depending on the Outpulse Without Tone settings administered for the system.
Port
Available only when the music on hold type is port.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks).
01 to 31 1 to 250 V1 to V9 01 to 31
Usage First and second characters are the cabinet number. Third character is the carrier Fourth and fifth character are the slot number. Sixth and seventh characters are the circuit number
Gateway Module Circuit
Protocol for Caller ID Analog Terminals
Determines the protocol or tones sent to a Caller ID telephone. The provider should have the correct protocol for each country.
Valid Entry Bellcore
V23Bell
Usage
Telcordia Technologies protocol with 212 modem protocol tones. Typically used in the U.S.
Telcordia Technologies protocol with V.23 modem tones. Typically used in Bahrain.
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Feature-related system parameters: page 1
Self Station Display Enabled
Enables or disables the displaying of the primary extension associated with a digital display telephone when the inspect button is pressed or when a feature access code (FAC) is entered.
· When using an FAC, the display continues until the user picks up the telephone or receives an incoming call.
· When using the inspect button, the display continues until the user presses the normal or exit button, or until the user picks up the telephone or receives an incoming call.
Trunk-to-Trunk Transfer
Regulations in some countries control the settings for this field. For more information, go to the Avaya Support website at http://support.avaya.com/ for current documentation, product notices, knowledge articles.
Valid Entry all
restricted none
Usage
Enables all trunk-to-trunk transfers. With this, telephone users can set up trunk-totrunk transfer, go on-hook without disconnecting the call, and forward the call to a remote location. This value is required for SIP support.
Restricts all public trunks (CO, WATS, FX, CPE, DID, and DIOD) from being transferred.
Restricts all trunks (except CAS and DCS) from being transferred.
Type
Indicates whether the source for Music on Hold is an announcement extension, an audio group, or a port on a VAL board.
Available only if music is administered as the treatment for calls placed on hold.
Valid Entry ext
group
port
Usage
Extension. Requires entry of the corresponding extension number of the announcement or audio source.
Analog group. Requires entry of the corresponding Music-on-Hold analog group number.
Analog/aux-trunk. Requires entry of the corresponding location of the Music-on-Hold analog or aux-trunk source.
Note:
A source identifier (extension number, audio group number, or port number) must be administered with the source type.
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Feature-related system parameters
Feature-related system parameters: page 2
LEAVE WORD CALLING PARAMETERS
Default Multimedia Outgoing Trunk Parameter Selection
Valid Entry 2x56 2x64
Usage Sets the default parameter for bandwidth and bearer for all video calls.
Related links Multimedia Call Access Code on page 279
Enhanced Abbreviated Dial Length (3 or 4)
Enhanced abbreviated dial lists are for users who need to store a vast quantity of speed-dialed numbers. The administrator might not be able to use all entry slots because of system capacity constraints.
Valid Entry 3 4
Usage Administrator can use 1000 enhanced list entries. Administrator can use 10,000 entries available.
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Feature-related system parameters: page 2
Maximum Number of External Calls Logged Per Station
When an external call is unanswered, the server running Communication Manager keeps a record of up to 15 calls, provided information on the caller identification is available, and the telephone message lamp lights. The telephone set displays the names and numbers of unsuccessful callers.
Valid Entry 0 to 15
Usage
The maximum number of calls that can be logged for each user. The assigned number cannot be larger than the entry in Maximum Number of Messages Per Station.
Maximum Number of Messages Per Station
Valid Entry 0 to 125
Usage
The maximum number of Leave Word Calling (LWC) messages that can be stored by the system for a telephone at a given time. With LWC, internal system users can leave a short pre-programmed message for other internal users. When the message is stored, the Automatic Message Waiting lamp lights on the called telephone.
Message Waiting Indication for External Calls
Enables or disables a station's ability to receive a message waiting indication when external calls are logged.
Prohibit Bridging Onto Calls with Data Privacy
Valid Entry y
n
Usage
Prohibits calls from getting bridged by any party, including Service Observing, Intrusion, Verify, and Bridging.
Allows calls to be bridged.
Stations With System-wide Retrieval Permission (enter extension)
The server running Communication Manager refers to extensions with system-wide retrieval permission as "super-retrievers".
Valid Entry Assigned extension
attd
Usage
Up to ten telephone extension numbers that have permission to retrieve LWC Messages or External Call Log records for all other telephones. A VDN extension is disallowed.
All attendants have retrieval permission.
Note:
An extension must be removed from this list before the station can be removed from the system.
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Feature-related system parameters
Feature-related system parameters: page 3
TTI/PSA PARAMETERS
CPN, ANI for Dissociated Sets
Specifies the ISDN calling party number (CPN), R2-MFC ANI, and CAMA CESID applied to calls made from PSA dissociated sets if no system-wide calling party information has been administered for those protocols. A dissociated set is a telephone without an extension. Available only if a default COR has been administered for dissociated sets. Accepts up to 20 digits.
Customer Telephone Activation (CTA) Enabled
Enables or disables the Customer Telephone Activation (CTA) feature that associates a physical telephone with a station extension. CTA applies only to DCP and analog telephones.
Default COR for Dissociated Sets
Available only when Terminal Translation Initialization is enabled for the system.
Valid Entry 0 to 995 blank
Usage
The Class of Restriction (COR) that the system uses for calls made from dissociated telephones. A dissociated set is a telephone without an extension.
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Feature-related system parameters: page 3
Enhanced PSA Location/Display Information Enabled
Enables or disables the display of Personal Station Access (PSA) information used to disassociate and associate telephones. When enabled, displays:
· PSA login and associated station information when a station is PSA associated · PSA logout and the port when a station is PSA dissociated Available only when Terminal Translation Initialization (TTI) is enabled for the system.
Hot Desking Enhancement Station Lock
Administers the Hot Desking Enhancement (HDE) feature. "Hot desking" is a feature that can lock and unlock telephones or move a fully customized station profile to another compatible telephone.
Valid Entry y n
Usage Enables the Hot Desking Enhancement feature. Disables the Hot Desking Enhancement feature.
Terminal Translation Initialization (TTI) Enabled
Technicians use the Terminal Translation Initialization (TTI) Enabled field to associate and disassociate telephones. This field is available only if TTI is enabled.
Caution:
Before making changes to the TTI settings, go to the Avaya Support website at http:// support.avaya.com/ for current documentation, product notices, knowledge articles.
Valid entry y
n
Usage
Starts ACTR, TTI, and PSA transactions for extension and telephone moves between ports.
Removes existing TTI port translations and prevents the generation of new TTI port translations.
TTI Security Code
Number that a TTI user uses to access TTI from a telephone or data terminal. Accepts up to seven digits. Available only if Terminal Translation Initialization (TTI) is enabled.
TTI State
The type of port translation used for unadministered digital ports. Available only when Terminal Translation Initialization (TTI) is enabled.
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Feature-related system parameters
Valid Entry data resume suspend voice
Usage
A stand-alone data module is the TTI port translation for the system. The activation and deactivation sequence is entered at the data terminal.
TTI is available after TTI has been manually suspended. The state of TTI returns to the state that it was in before TTI was manually suspended.
TTI voice or TTI data translations are temporarily unavailable. The system does not remove existing TTI translations.
A voice or voice/data terminal is the TTI port translation for the system. The activation and deactivation sequence is entered from a telephone.
Unnamed Registrations and PSA for IP Telephones
Use this field to allow IP telephones access to the Personal Station Access (PSA) feature. If you set this field to y, IP telephones can register into the PSA dissociated state. The PSA dissociated state is also known as TTI unmerged state, TTI state, and Unnamed Registered for H.323 standards state.
EMU PARAMETERS
EMU Inactivity Interval for Deactivation (hours)
A system-wide administrable interval for EMU de-registration at the visited switch. This timer is applicable to inter and intra-Communication Manager EMU registrations.
Valid Entry 1 to 24
blank
Usage
The interval in hours, after which a visiting user is dropped due to inactivity. Default is 1. An entry of 1 means that after 1 hour of inactivity, the telephone is dropped from the visited home server.
The timer is not used and the visited station remains active until de-registration by another means occurs.
CALL PROCESSING OVERLOAD MITIGATION
Restrict Calls
Indicates the type of calls to block first during overload traffic conditions on the system.
Valid Entry stations-first
all-trunks-first public-trunks-first
Usage
Deny new traffic generated by internal stations, allowing inbound calls only. This works best in call center environments.
Deny all incoming trunk calls and does not block station off-hook.
Deny incoming calls on ISDN public trunks and analog CO trunks.
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Feature-related system parameters: page 4
Call Pickup on Intercom Calls
Provides or prohibits the system-wide use of Call Pickup or Directed Call Pickup features on intercom calls.
Controlled Outward Restriction Intercept Treatment
The type of intercept treatment the caller receives when the call is outward restricted.
Valid Entry announcement
attendant
Usage
Provides a recorded announcement to calls that cannot be completed as dialed. The calling party receives indication that the call is receiving Intercept Treatment. Requires an extension number for the announcement in the associated field.
Note:
If you enter a shortened extension of the multilocation dial plan, the system does not perform certain administration and validation tasks. Therefore, the system might not display the resultant warnings or submittal denials.
Allows attendants to provide information and assistance to outgoing calls that cannot be completed as dialed or that are transferred to incomplete or restricted stations.
Table continues...
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Feature-related system parameters
Valid Entry extension
tone
Usage
Routes to an extension. Requires an extension number for the extension in an associated field. Cannot be a VDN extension.
Provides a siren-type tone to internal calls that cannot be completed as dialed.
Controlled Station-to-Station Restriction
The type of intercept treatment the caller receives when the call is placed to a restricted telephone.
Valid Entry announcement attendant extension
tone
Usage Provides an announcement. Requires entry of the announcement extension. Intercepted calls are redirected to the attendant. Routes the call to an alternate extension. Requires entry of the station or individual attendant. This cannot be a VDN extension. Intercepted calls receive intercept (siren) tone.
Controlled Termination Restriction (Do Not Disturb)
The type of intercept treatment the caller receives when the call is placed to a termination restricted telephone.
Valid Entry announcement
attendant coverage extension
tone
Usage
Redirects intercepted calls to an announcement. Requires an associated extension number.
Redirects intercepted calls to the attendant.
Redirects intercepted calls to coverage.
Redirects intercepted calls to an extension. Requires an associated extension number. Cannot be a VDN extension.
Provides a siren-type tone to calls that cannot be completed as dialed.
Deluxe Paging and Call Park Timeout to Originator
Enables or disables the Call Park feature so that a user can retrieve a call that is on hold from any other telephone within the system. For example, a user can answer a call at one extension, put the call on hold, and then retrieve the call at another extension. Or the user can answer a call at any telephone after an attendant or another user pages the user. Paged calls that are to be parked require separate activation of the Call Park feature.
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Valid Entry y n
Feature-related system parameters: page 4
Usage The system redirects a parked call to the user who parked the call. The system redirects a parked call to the attendant.
Drop Parking User From the Call After Timeout
Valid Entry y n
Usage Drops the parking user from the call automatically after the timeout. Does not drop the parking user from the call automatically after the timeout.
Emergency Access Redirection Extension
The assigned extension number or Vector Directory Number (VDN) where emergency queue overflow redirects.
Extended Group Call Pickup
Enables call pickup groups to answer calls directed to another call pickup group.
Valid Entry flexible
simple
none
Usage
Flexible feature version supporting a one-to-n (pickup group-to-extended pickup group) mapping.
Simple feature version with a one-to-one pickup group-to-extended pickup group mapping supported.
Extended group call pickup is ignored.
Number of Emergency Calls Allowed in Attendant Queue
Valid Entry 0 to 75
Usage
The number of emergency calls allowed in the attendant queue before additional calls are routed to the backup extension.
Reserved Slots for Attendant Priority Queue
Valid Entry 2 to 342
Usage The number of calls that can go in to the emergency queue.
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Feature-related system parameters
Time Before Off-Hook Alert
Valid Entry 1 to 3000
Usage
The time in seconds that a telephone with an Off-Hook Alert Class of Service can remain off-hook (after intercept tone has started) before an emergency call is sent to the attendant.
AUTHORIZATION CODE PARAMETERS
Attendant Time Out Flag
Applies only to remote users or incoming calls over trunks requiring an authorization code. Available only if authorization codes are enabled for the system.
Valid Entry y
n
Usage
A call is routed to the attendant if the caller does not dial an authorization code within 10 seconds or dials an invalid authorization code.
The caller receives intercept tone.
Related links Authorization Codes Enabled on page 299
Authorization Code Cancellation Symbol
The symbol a caller must dial to cancel the 10-second wait period during which the user can enter an authorization code. Available only if authorization codes are enabled for the system.
Valid Entry (#) (1)
Usage The main and tandem servers or switches are both of the same type. An Avaya server or switch is part of the complex or network.
Authorization Code Length
Available only if authorization codes are enabled for the system. Security alert: Enhance system security by using the maximum length for your authorization code.
Valid Entry 4 to 13 digits
Usage
The required length of the authorization code. All administered authorization codes must be this length.
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Authorization Codes Enabled
Enables or disables authorization codes on a system-wide basis. Available only if authorization codes are enabled.
Security alert: To maintain system security, use authorization codes.
Controlled Toll Restriction Intercept Treatment
Applies an intercept treatment to a toll call during the call processing. Available only if the Controlled Toll Restriction feature is activated.
Valid Entry announcement
attendant extension
tone
Usage
Intercepted calls are redirected to an announcement. Requires an assigned announcement extension.
Intercepted calls are redirected to the attendant.
Intercepted calls are redirected to an extension. Requires an extension assigned to a station or individual attendant.
Intercepted calls receive intercept (siren) tone.
Controlled Toll Restriction Replaces
Activates the Controlled Toll Restriction feature for the customer to use additional types of calling restrictions on guest room telephones.
Valid Entry outward station-station
termination total toll
Usage Guests cannot place calls to the public network. Guests cannot place or receive calls between guest rooms or administrative staff voice terminals. Guests cannot receive any calls. Guests cannot place or receive any calls. Guests cannot place toll calls, but can place local free calls.
Display Authorization Code
Enables or disables displaying authorization code digits on the set during dialing. Applies only to DCP, not to BRI or hybrid sets.
Security alert: To enhance system security, do not display authorization code digits during dialing.
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Feature-related system parameters
Feature-related system parameters: page 5
SYSTEM PRINTER PARAMETERS
The system printer is the printer dedicated to support scheduled reports.
Endpoint
Valid Entry SYS_PRNT
blank
Usage
If the system printer is connected over a TCP/IP link, and the link is defined as SYS_PRNT for IP Services
Not administered
Lines Per Page
Valid Entry 24 to 132
Usage The number of lines per page required for the report.
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SYSTEM-WIDE PARAMETERS
COR to Use for DPT
The Class of Restriction used for the Dial Plan Transparency (DPT) feature.
Valid Entry station
unrestricted
Usage
The Facility Restriction Level (FRL) of the calling station determines whether that station is permitted to make a trunk call and if so, which trunks it is eligible to access. This is the default.
The first available trunk preference pointed to by ARS routing is used.
Emergency Extension Forwarding (min)
If an emergency call disconnects, public safety personnel always attempt to call back. If the ELIN that was sent differs from the caller's extension number, the return call rings a set other than the one that dialed 911. To overcome this limitation, you can automatically forward that return call to the set that placed the emergency call for an administered period of time.
Emergency Extension Forwarding only applies if the emergency location extension number is an extension on the same switch as the extension that dialed 911. Customers who have several switches in a campus should assign emergency location extensions accordingly.
Sets the Emergency Extension Forwarding timer for all incoming trunk calls if an emergency call gets cut off.
Valid Entry 0 to 999
Usage
The time in minutes that an incoming trunk call forwards to the extension that made the initial 911 call. The default value for both new installs and upgrades is 10.
Note:
If a user at the emergency location extension (the extension that made the initial 911 call) manually turns off the Call Forwarding feature, the feature is off no matter how many minutes remain on the timer.
Enable Dial Plan Transparency in Survivable Mode
Enables or disables Dial Plan Transparency (DPT) without changing or removing other feature administration associated with DPT. DPT is enabled if a gateway registers with a Survivable Remote Server (Local survivable processor), or a port network registers with a Survivable Core Server (Enterprise Survivable Server).
Enable Inter-Gateway Alternate Routing
Enables or disables the Inter-Gateway Alternate Routing (IGAR) feature that provides a means of alternately using the public network when the IP-WAN is incapable of carrying the bearer connection.
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Feature-related system parameters
IGAR Over IP Trunks
The system displays the field when you set the Enable Inter-Gateway Alternate Routing field to y.
Valid Entry allow
skip
Usage
The IGAR feature uses H.323 or SIP trunks when selecting a trunk from a Route Pattern.
The IGAR feature skips H.323 and SIP trunks when selecting a trunk from a Route Pattern. By default, the value is skip.
Switch Name
A name used for identifying the switch. Accepts up to 20 alphanumeric characters.
MALICIOUS CALL TRACE PARAMETERS
Apply MCT Warning Tone
Enables or disables an audible tone to the controlling station when a Malicious Call Trace (MCT) recorder is actively recording a malicious call.
Delay Sending Release (seconds)
Available only if Malicious Call Trace is enabled for the system.
Valid Entry 0 to 30
Usage
The time in seconds the system waits before sending an ISDN release message in response to receiving an ISDN disconnect message. Available in increments of 10.
MCT Voice Recorder Trunk Group
Valid Entry 1 to 2000
Usage Assigns the trunk group number for Malicious Call Trace (MCT) voice recorders.
SEND ALL CALL OPTIONS
Auto Inspect on Send All Calls
Valid Entry y
n
Usage
The system displays Calling Party information for calls that are silently alerting their station because of the Send-All-Calls feature.
Calling Party display for calls sent directly to Coverage by the Send-All-Calls feature is not guaranteed.
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Send All Calls Applies to
Valid Entry station
extension
Usage
Any call to a station, regardless of the number dialed, causes calls to that station's own extension to be sent immediately to Coverage, or causes calls to different extensions assigned to the station as bridged appearances to have Ring-Ping notification if redirect notification is enabled.
Only the calls sent to that extension are placed to coverage.
UNIVERSAL CALL ID
Create Universal Call ID (UCID)
Enables or disables the generation of a UCID for each call when necessary.
UCID Network Node ID
Valid Entry 1 to 327 blank
Usage
A number unique to this server or switch in a network of switches. This number is an important part of the UCID tag and must be unique to the server or switch.
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Feature-related system parameters
Feature-related system parameters: page 6
7405ND Numeric Terminal Display
Enables or disables allowing a 7405ND type of station. This is not an actual telephone type, but it can be used to define ports for certain types of voice messaging systems. This numeric display setting sends only numbers, and not names, to the messaging system. Related links Type on page 758
7434ND
Enables or disables allowing a 7434ND type of station. This is not an actual telephone type, but it can be used to define ports for certain types of messaging systems. Used if the voice messaging system operates in Bridged Mode. Related links Type on page 758
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Allow AAR/ARS Access from DID/DIOD
Enables or disables allowing calls for DID and DIOD type trunk groups to complete calls using ARS or AAR.
Allow ANI Restriction on AAR/ARS
(For Russia only). Enables or disables allowing a call placed over a Russian shuttle trunk or a Russian rotary trunk using AAR or ARS to have the ANI requirement administered as restricted. In this case, when ANI is requested, if the request fails, the call immediately drops. Related links ANI Reqd on page 19 ANI Reqd on page 24 Request Incoming ANI (non-AAR/ARS) on page 607
Attendant Tone
Enables or disables providing call progress tones to the attendants.
Auto Hold
Enables or disables the Automatic Hold feature on a system-wide basis.
Auto Start
If enabled, the Start buttons on all attendant consoles are disabled and the Automatic Start feature is enabled.
Bridging Tone
Enables or disables providing a bridging tone when calls are bridged on primary extensions.
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Feature-related system parameters
Conference Parties without Public Network Trunks
Valid entry 3 to 6, or 12
Usage The maximum number of parties allowed in a conference call involving no public network trunks.
Note:
For 12 parties to participate in a conference, you must enable the 12party Conferences field in the Feature-Related System-Parameters screen.
Conference parties with public network trunks
Available only if public network trunks are allowed on a conference call.
Valid Entry 3 to 6, or 12
Usage The maximum number of parties allowed in a conference call involving a public network subscriber.
Note:
For 12 parties to participate in a conference, you must enable the 12party Conferences field in the Feature-Related System-Parameters screen.
Related links Public network trunks on conference call on page 308
Conference Tone
Enables or disables providing a conference tone as long as three or more calls are in a conference call.
Note:
Bridging and Conference Tones are not supported by all countries. If these tones are enabled for countries other than Italy, Belgium, United Kingdom, or Australia, the tones are equivalent to no tone (silence) unless the tone is independently administered or customized.
Related links Tone Generation on page 896
DID Busy Treatment
Specifies how to handle a direct inward dialing (DID) call to a busy station.
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Valid Entry attendant tone
Usage Call is routed to attendant. Caller hears a busy tone.
Feature-related system parameters: page 6
DTMF Tone Feedback Signal to VRU - Connection, Disconnection
Available only if DTMF Feedback Signals for the Voice Response Unit (VRU) are enabled for the system.
Valid Entry 0 to 9, *, #, A, B, C, D
blank
Usage
The code used to connect or disconnect the VRU. This can be a single digit, or a combination such as *99 to connect, #99 to disconnect. The tones must be programmed at the VRU as well.
No tone is sent to the VRU.
Related links DTMF Feedback Signals For VRU on page 843
Intrusion Tone
Enables or disables applying an intrusion tone (executive override) when an attendant intrudes on the call.
Line Intercept Tone Timer (seconds)
Valid Entry 0 to 60
Usage
How long an analog station user can wait after hearing warning tone without going on hook, before the station is placed in the lockout state.
Long Hold Recall Timer (seconds)
Valid Entry 0 to 999
Usage
The number of seconds a call can be on hold before the system re-alerts the user to remind them of the call.
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Feature-related system parameters
Mode Code Interface
Activates or deactivates the use of the Mode Code Voice Mail System Interface to connect the server running Communication Manager over a DTMF interface to other voice mail systems.
Note:
After making changes, log off and log back on to access the Mode Code Related System Parameters.
Night Service Disconnect Timer (seconds)
Valid Entry 10 to 1024 blank
Usage
The number of seconds a trunk call can be unanswered during night service before being disconnected. The trunk must not have Disconnect Supervision for this timer to apply.
Public network trunks on conference call
Valid Entry 0 to 5, or 11
Usage The number of public network trunks allowed on a conference call.
Note:
For 12 parties to participate in a conference, you must enable the 12party Conferences field in the Feature-Related System-Parameters screen.
Recall from VDN
Indicates whether or not a call that is transferred to a VDN and then routed to a station is recalled to the originating station after the Station Call Transfer Recall Timer expires. If enabled, calls are recalled from a VDN when the Station Call Transfer Recall Timer expires.
Available only if Basic Vectoring and Vectoring (Prompting) are enabled for the system.
Related links Vectoring (Basic) on page 845 Vectoring (Prompting) on page 846 Station Call Transfer Recall Timer (seconds) on page 309
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Feature-related system parameters: page 6
Reset Shift Timer (seconds)
Valid Entry 0 to 255
Usage
Specifies the number of seconds that reset shift dial tone is audible before busy tone is heard. With reset shift dial tone, the user can dial a new extension by dialing one new digit that replaces the last digit of the extension previously dialed. The new digit replaces the last digit of the extension previously dialed. An entry of 0 disables this feature.
Used only for station-to-station calls or private network calls using ISDN trunks.
Short Interdigit Timer (seconds)
Valid Entry 3 to 9
Usage
The time limit that digit analysis will wait for the next digit when it has predicted that all the digits have already been collected.
Special Dial Tone
If enabled, Special Dial Tone can be used. Special dial tone notifies an analog-telephone user if certain features are still active when the user goes off-hook. These features include:
· Call Forwarding · Send All Calls · Do Not Disturb Requires a TN2182 circuit pack, or any newer tone generation resource. Related links Special Dial Tone on page 350
Station Call Transfer Recall Timer (seconds)
A user-transferred call (station-to-station, a trunk call, or a DCS call) can re-terminate with priority ringing back to the station user who initiates the transfer operation if the transfer-to party does not answer the call within the administered Station Call Transfer Recall timer.
Valid Entry 0 to 999
Usage
The time in seconds before a call redirects back to the station user who initiated the transfer operation. The entry 0 disables this feature.
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Feature-related system parameters
Trunk Alerting Tone Interval (seconds)
Valid Entry 1 to 60
Usage
Specifies the interval at which the alerting tone is repeated on the call. By default, the value is 15 seconds.
Related links Outgoing Trunk Alerting Timer (minutes) on page 132
Unanswered DID Call Timer (seconds)
Valid Entry 10 to 1024
blank
Usage
Limits how long a DID call can remain unanswered before routing to the DID/TIE/ ISDN Intercept Treatment. This timer interacts with the nonadministrable 50-second Wait for Answer Supervision Timer (WAST). The WAST timer overrides this field. Therefore, if this field is set to a value equal to or greater than 50 seconds, the caller receives intercept tone instead of the normal attendant or announcement treatment that is given when the Unanswered DID Call Timer expires before the WAST. If the Unanswered DID Call Timer expires while the DID call is being processed by call vectoring, the timer is ignored.
Disables the timer.
Related links Wait Answer Supervision Timer on page 328
Use Trunk COR for Outgoing Trunk Disconnect/Alert
Indicates whether the outgoing trunk disconnect timer or the outgoing trunk alerting timer is set based on the COR of the originating station or the trunk group. If enabled, the timer is based on the COR of the trunk, not the originating station. By default, this field is disabled.
Important:
You must disable outgoing trunk disconnect and outgoing trunk alerting timers at the same time. If you try to administer both timers, the system displays the following message:
Cannot enable both Outgoing Trunk Disc and Outgoing Trunk Alert timers
Related links Outgoing Trunk Disconnect Timer (minutes) on page 132 Outgoing Trunk Alerting Timer (minutes) on page 132
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Feature-related system parameters: page 6
DISTINCTIVE AUDIBLE ALERTING
Available only if Tenant Partitioning is disabled for the system.
Attendant Originated Calls
Indicates which type of ringing applies to attendant-originated calls.
Valid Entry internal external priority
Usage Internal ringing applies to attendant-originated calls. External ringing applies to attendant-originated calls. Default is external. Priority ringing applies to attendant-originated calls.
Distinctive Audible Alerting (Internal, External, Priority)
The number of rings for Internal, External, and Priority calls. For virtual stations, this applies to the mapped-to physical telephone.
Valid Entry 1 2 3
Usage One burst of ringing signal per period. Default for internal calls. Two bursts of ringing signal per period. Default for external and attendant calls. Three bursts of ringing signal per period. Default for priority calls.
Related links Distinctive Audible Alerting (Internal, External, Priority) on page 873
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Feature-related system parameters
Feature-related system parameters: page 7
Important: To enable 12-party conference, you must disable the DCS feature. DCS field is available in the change system-parameters customer-options page.
CONFERENCE/TRANSFER
12-party Conferences
If enabled, 12 parties can participate in ad-hoc and meet-me conferences. If disabled, the maximum number of participants in the conference remains 6.
Abort Conference
DCP, hybrid, IP, wireless, or ISDN-BRI telephone users can abort the conference operation when they are disconnected. If enabled, this field changes a call placed on soft-hold in the conferencepending status to hard-held status if the user disconnects.
Abort Transfer
If enabled, stops the transfer operation whenever users press a non-idle call appearance button in the middle of the transfer operation, or when they disconnect. If the system is configured to
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Feature-related system parameters: page 7
transfer calls upon disconnecting the call, users can press the Transfer button, dial the complete transfer-to number, and disconnect to transfer the call. Users must select another non-idle call appearance to abort the transfer. Requires DCP, Hybrid, IP, ISDN-BRI or wireless telephones.
External Ringing for Calls with Trunks
Specifies ringing behavior on external trunk calls that are transferred or conferenced by stations or attendants, or extended by the attendant to an on-switch extension.
Valid Entry all-calls local-only none remote-only
Usage
All external trunk calls that are transferred or conferenced (either locally or remotely) receive external ringing.
External trunk calls that are transferred or conferenced locally receive external ringing.
External ringing does not apply to external trunk calls that are transferred or conferenced.
External trunk calls that are transferred or conferenced remotely receive external ringing. This is the default value.
Maximum Ports per Expanded Meet-me Conf
Available only if ports are administered for the Expanded Meet-me Conference feature.
Valid Entry 3 to 300
Usage
The maximum number of conferees in an Expanded Meet-me Conference. This is a system-wide limit (that is, not administrable on a per Expanded-Meet-me VDN basis).
No Dial Tone Conferencing
If enabled, eliminates dial tone while setting up a conference when another line is on hold or is alerting.
No Hold Conference Timeout
Valid Entry 20 to 120
Usage
The number of seconds No Hold Conference call setup times out. The system Answer Supervision timer should be set to a value less than this.
Select Line Appearance Conferencing
User can use the line appearance rather than the Conference button to include a call in a conference. If a user is on a call, and another line is on hold or an incoming call alerts on another line, the user can press the Conference button to bridge the calls together. Using the select line appearance capability, the user can press a line appearance button to complete a conference instead of pressing the Conference button a second time.
Enabling this field activates Select Line Appearance Conferencing.
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Feature-related system parameters
Transfer Upon Hang-Up
With this feature, users can transfer a call by pressing the Transfer button, dialing the required extension, and then disconnecting the active call. Users using the Call Park FAC can park a call without pressing the Transfer button a second time.
Note: Communication Manager Release 6.2 and earlier supported this feature on H.323 and DCP stations. Communication Manager Release 6.3.6 extends this support to 96x1 SIP stations. However, for SIP stations, this field works only when the Team button feature is enabled.
Unhold
With unhold, the user can press the hold button on a telephone to release a hold (if no other line appearance is on hold or alerting). This does not apply to BRI telephones or attendant consoles. Enabling this field activates this unhold capability.
ANALOG BUSY AUTO CALLBACK
If the Analog Busy Auto Callback Without Flash (ACB) feature is enabled, when a caller places a call through an analog station, and the called station is busy and has no coverage path or forwarding, then an announcement plays, announcing that the station is busy and prompting the caller to enter 1 for ACB or 2 to cover to a voice mail hunt group extension.
Announcement
The extension of the announcement that plays for the Analog Busy Auto Callback Without Flash feature. This field cannot be left blank. Available only if Auto Callback Without Flash is enabled.
Voice Mail Hunt Group Ext
A voice mail hunt group extension where the call is forwarded if the user enters 2 at the ACB announcement prompt. Available only if Auto Callback Without Flash is enabled.
Without Flash
If enabled, provides automatic callback for analog stations without flashing the hook. It is applied only when the called station is busy and has no other coverage path or call forwarding. The caller can enable the automatic callback without flashing the hook or entering the feature access code.
AUDIX ONE-STEP RECORDING
On stations administered with this feature button, users can activate and deactivate the recording of active calls to their voice messaging system with the press of one button.
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Feature-related system parameters: page 7
Apply Ready Indication Tone To Which Parties In The Call
Use this field to administer who hears the ready indication tone.
Valid entry all initiator
none
Usage
All parties in the call hear the tone. The default value is all.
Only the initiator hears the tone. The initiator is the user who activates the Audix one-step recording feature.
None of the parties in the call hear the tone.
Note:
When the Apply Ready Indication Tone To Which Parties In The Call field is set to initiator, all parties in the call hear the ready indication tone. When the Apply Ready Indication Tone To Which Parties In The Call field is set to all and the Interval For Applying Periodic Alerting Tone (seconds) field is set to 0, the parties in the call do not hear the ready indication tone and the alerting tone.
Interval For Applying Periodic Alerting Tone (seconds)
Available only if all parties on a call hear the voice messaging recording ready tone.
Valid Entry 0 to 60
Usage
The number of seconds between alerting tones. The default value is a 15 second interval. Zero disables the tone.
Recording Delay Timer (msecs)
Valid Entry
0 to 4000 in increments of 100
Usage The delay interval before starting a voice messaging recording.
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Feature-related system parameters
Feature-related system parameters: page 8
ISDN PARAMETERS
Delay for USNI Calling Name for Analog Caller ID Phones (seconds)
For analog caller ID telephones, Communication Manager waits for the FACILITY message (includes the caller information) before delivering an incoming call to the receiver.
Valid Entry 0 to 6
Usage
Communication Manager waits for the administered duration for the FACILITY message before delivering the incoming call to the receiver. The default value of the field is 0 which indicates that the feature is disabled. Use the timer value of 3 seconds.
Note: On page 3 of the ISDN Trunk Group screen, you must set US NI Delayed Calling Name Update field to y for this feature to work.
Display Connected Name/Number for ISDN DCS Calls
If enabled, displays the connected name/number (if received) for ISDN DCS calls.
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Feature-related system parameters: page 8
Feature Plus Ext
An extension used for proper termination of Feature Plus signaling. For example, Message Waiting Indication (MWI) requires this extension to send the indication to the appropriate server running Communication Manager. Available only if ISDN Feature Plus signaling is enabled for the system.
International CPN Prefix
You can use this number to apply prefixes to international calling numbers for display at receiving telephones. Accepts up to five digits and includes the characters * and #. This number is useful for those telephones that use or implement call back features based on incoming call numbers.
When an ISDN-PRI call arrives, the incoming call setup is analyzed for:
· Whether the Type of Address (TOA) is national or international
· Whether the Numbering Plan Identifier (NPI) is Unknown or ISDN/Telephony
This administered prefix is applied to international calls.
Prefixing applies to any subsequent display on the same server when the call is transferred, covered, or forwarded. The same prefixing applies to outgoing ISDN-PRI calls when the connected number information is returned and meets the same TOA and NPI criteria. The prefix plus the calling/connected number digit string is limited to 15 digits, with truncation occurring at the least significant digits.
Maximum Length
Available only if unknown numbers are administered as internal for the voice messaging system.
Valid Entry 1 to 20
Usage
The maximum length of an unknown private number. Any unknown number longer than the administered value is external. This field cannot be blank.
MWI - Number of Digits Per Voice Mail Subscriber
The number of digits per voice messaging system subscriber. Available only if Basic Supplementary Services or ISDN Feature Plus signaling is enabled for the system.
Valid Entry 3 to 7
Usage
The digit string length of subscribers translated in the Message Center entity. The value in this field must match the value administered for the voice messaging system.
National CPN Prefix
Applies prefixes to national calling numbers for display at receiving telephones. This is useful for those telephones that use or implement call back features based on incoming call numbers. When an ISDN-PRI call arrives, the incoming call setup is analyzed for: (1) whether the Type of Address (TOA) is national or international, and (2) whether the Numbering Plan Identifier (NPI) is Unknown or ISDN/Telephony.
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Feature-related system parameters
This administered prefix is applied to national calls. Prefixing applies to any subsequent display on the same server when the call is transferred, covered, or forwarded. The same prefixing applies to outgoing ISDN-PRI calls when the connected number information is returned and meets the same TOA and NPI criteria. The prefix plus the calling/connected number digit string is limited to 15 digits, with truncation occurring at the least significant digits.
Valid Entry 0 to 9 * or # blank
Usage
A five-digit number that you can apply as prefixes to national calling numbers for display.
Pass Prefixed CPN to ASAI
If enabled, passes Calling Party Number information (CPN) to ASAI. The prefixed number is not passed on to other adjuncts, Call Detail Recording, or servers/switches.
Path Replacement While in Queue/Vectoring
Use this field to allow Path Replacement after queue/vector processing has started. Depending on the version of CMS, some calls can go unrecorded if this capability is enabled. For more information, go to Avaya Support website at http://support.avaya.com/.
Path Replacement with Measurements
Use this field to allow QSIG path replacement or DCS with Reroute to be attempted on measured calls.
QSIG/ETSI TSC Extension
The phantom endpoint extension for QSIG Call Independent Signaling Connections (CISCs) that are similar to NCA Temporary Signaling Connections (TSCs) (both incoming and outgoing). ETSI protocol TSCs as well as QSIG TSCs are supported.
QSIG Path Replacement Extension
The extension for the system used as part of the complete number sent in the Path Replacement Propose message.
Send Custom Messages Through QSIG?
If enabled, provides appropriate display information over QSIG links. Display information can include the Posted Messages feature.
Send ISDN Trunk Group Name on Tandem Calls
If enabled, provides consistent display information regardless of trunk type. Also provides only trunk group name.
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Feature-related system parameters: page 8
Send Non-ISDN Trunk Group Name as Connected Name
If enabled, sends a name of the non-ISDN trunk group as the connected name when a call routes from ISDN to non-ISDN and the call is answered.
Unknown Numbers Considered Internal for AUDIX
Controls the treatment of an ISDN number whose numbering plan identification is "unknown" in a QSIG centralized voice messaging system arrangement. Available only if ISDN trunks are enabled for the system and if the hunt group is administered to send the calling party number to the voice messaging system.
Valid Entry y
n
Usage
The unknown number is "internal" and the voice messaging system tries to find a calling party name match for the digit string. If a name match is found, the voice messaging system provides the calling party's name. If no name is found, the voice messaging system provides the calling party's telephone number.
The unknown number is "external" and the voice messaging system provides the calling party's telephone number.
USNI Calling Name for Outgoing Calls?
Valid Entry y n
Usage Sends a name on outgoing calls over NI PRI trunks. Prevents sending calling name information with outgoing calls over NI PRI trunks. Overrides the trunk group Send Name if enabled.
Caution:
The service provider's local telephone company central office (CO) must be capable of accepting calling name information from Communication Manager in this way. For example, if the CO has a 5ESS, it must be a generic 5EXX or later. Failure to validate the CO capability might cause the CO to drop outgoing calls from the Avaya S8XXX Server. In this case, this field should be disabled.
PARAMETERS FOR CREATING QSIG SELECTION NUMBERS
Level 1 Code
The first level regional code of the Avaya S8XXX Server in the network.
Administer this field carefully. Communication Manager does not verify if you have entered a code that supports your entry in the Network Level field. Accepts up to five digits. Because blank regional codes are valid, an entry is not required if the Network Level field is 1 or 2.
In QSIG standards, this Level 1 code is called the Level 0 Regional Code.
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Feature-related system parameters
Available only if the Network Level is set to 1 or 2.
Level 2 Code
The second level regional code of the Avaya S8XXX Server in the network.
Administer this field carefully. Communication Manager does not check to ensure you have entered a code that supports your entry in the Network Level field. Accepts up to five digits. Because blank regional codes are valid, an entry is not required if the Network Level field is 2.
Available only if the Network Level is set to 2.
In QSIG standards, this Level 2 code is called the Level 1 Regional Code.
Network Level
The value of the highest regional level employed by the PNP network. Use the following table to find the relationship between the network level and the Numbering Plan Identification/ Type of Number (NPI/TON) encoding used in the QSIG PartyNumber or the Calling Number and Connected Number IEs.
Valid Entry 0 1 2 blank
Usage
NPI - PNP
TON - local
NPI - PNP
TON - Regional Level 1
NPI - PNP
TON - Regional Level 2
If this field is blank and the Send Calling Number and/or Send Connected Number field is y or r with private specified for the Numbering Format field on the ISDN Trunk Group screen, the Calling Number and/or Connected Number IEs is not sent. If the field is left blank but private has been specified in the Numbering Format field on the ISDN Trunk Group screen, the Identification Number (PartyNumber data type) is sent for QSIG PartyNumbers encoded in ASN.1-defined APDUs. In this case, the ASN.1 data type containing the PartyNumber (PresentedAddressScreened, PresentedAddressUnscreened, PresentedNumberScreened, or PresentedNumberUnscreened) is sent marked as PresentationRestricted with NULL for the associated digits.
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Feature-related system parameters: page 9
Feature-related system parameters: page 9
CPN/ANI/ICLID PARAMETERS
CPN/ANI/ICLID Replacement for Restricted Calls
Use this field to specify a text string to replace the restricted numbers on the telephone display. You can enter up to 15 characters.
Note: This field does not apply to SIP telephones.
CPN/ANI/ICLID Replacement for Unavailable Calls
Use this field to specify a text string to replace the unavailable numbers on the telephone display. You can enter up to 15 characters.
Note: This field does not apply to SIP telephones.
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Feature-related system parameters
DISPLAY TEXT
Extension only label for Team button on 96xx H.323 terminals
Available only if the Team Btn Display Name for the Class of Restriction is enabled.
Valid entries y
n
Usage
For 96xx H.323 telephones, displays the station extension without the team button label. 96xx H323 firmware version 2.0 or greater is recommended as it provides a special icon for team buttons. This field does not impact label customizations.
For 96xx H.323 telephones, displays the station extension with the team button label.
Identity When Bridging
Determines whether the telephone display shows the identity of the extension that the bridged appearance represents, or the identity of the physical telephone that the bridged appearance button is on. This applies when the bridged appearance is the caller, and when it is the answering party.
Note:
When you choose the station option, you must update the Public/Unknown Numbering Format with the Extension Codes of the stations that display the caller's or answering party's assigned identification.
Valid Entry principal station
Usage The extension that the bridged appearance button points to. This is the default. The physical telephone that the bridged appearance button is on.
User Guidance Display
If enabled, the telephone display shows user guidance messages. This field is disabled by default.
INTERNATIONAL CALL ROUTING PARAMETERS
Local Country Code
A valid PSTN E.164 country code. For example, an SBS node in the United States uses a code of 1. For a list of country codes, see the International Telecommunications Union List of ITU-T Recommendation E.164 Assigned Country Codes. Accepts up to three digits.
Related links SBS on page 530 International Access Code on page 323
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Feature-related system parameters: page 9
Group Type on page 508 Supplementary Service Protocol on page 521 Carrier Medium on page 505
International Access Code
The access code required by the PSTN to route calls out of the country. This code is included with the telephone number received from the SBS terminating node if the Local Country Codes of the originating and terminating nodes are different. For example, an SBS node in the United States uses an access code of 011. Accepts up to five digits.
Note:
Once administered, this field cannot be cleared until all trunk groups administered for SBS signaling have been removed. If the international call routing parameters are not administered for the system and SBS is enabled for the trunk, a warning is displayed: Must set INTERNATIONAL CALL ROUTING parameters on system-parameters features.
Related links SBS on page 530 Local Country Code on page 322 Group Type on page 508 Supplementary Service Protocol on page 521 Carrier Medium on page 505
SCCAN PARAMETERS
Enable Enbloc Dialing without ARS FAC
If the Enable Enbloc Dialing without ARS FAC field is enabled, Communication Manager inserts the ARS access code, when a dialed digit string has:
· Minimum Digit Length or more digits, and is sent from the telephone to Communication Manager in a single message
· a matching ARS entry
If the dialed digit string does not meet above points, the digit string is used as it is without prepending the ARS access code. The call is passed to ARS for digit analysis. By default, the Enable Enbloc Dialing without ARS FAC field is disabled.
Minimum Digit Length
Available only if Enbloc Dialing without an ARS FAC is enabled.
Valid Entry 1 to 20
Usage
The number of digits before Enbloc Calling Treatment is activated. Default is extension length plus 1.
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Feature-related system parameters
CALLER ID ON CALL WAITING PARAMETERS
Caller ID on Call Waiting Delay Timer (msec)
Valid Entry 5 to 1275
Usage The required delay in 5-millisecond intervals. Default is 200.
Feature-related system parameters: page 10
Allow Conference via Flash
If enabled, an analog station can use flash to conference calls.
Allow Crisis Alert Across Tenants
If enabled, allows crisis alert across different tenant partitions. Disabled by default.
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Feature-related system parameters: page 10
Charge Display Update Frequency (seconds)
Available only if Advice of Charge or Periodic Pulse Metering with display functions is used.
Valid Entry 10 to 60 blank
Usage
The amount of time in seconds between charge-display updates. Frequent display updates might have considerable performance impact. If the duration of a call is less than the Charge Display Update Frequency, the display does not automatically show charge information.
Date Format on Terminals
The system displays the Date Format on Terminals field if the Multinational Locations field on the System Parameters Customer Options screen is set to n. This field affects the 607/2400/4600/6400 terminals. The date is in mm/dd/yy, dd/mm/yy, and yy/mm/dd formats.
Edit Dialing on 96xx H.323 Terminals
If enabled, an end-user can dial a number when the telephone is on-hook. Disabled by default.
Hear Zip Tone Following VOA?
Enables or disables zip tone alerts to a telephone user when the announcement has completed and a caller is now connected. CallMaster set and attendant console users hear double zip tone following the announcement. All other telephone users hear single zip tone.
Note:
This field does not effect auto-answer zip tone heard prior to the VDN of Origin Announcement (VOA).
Intercept Treatment on Failed Trunk Transfers
Valid Entry y n
Usage Provides intercept treatment to calls failing trunk transfers. Drops calls failing trunk transfers.
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Feature-related system parameters
Level of Tone Detection
Used only when users are having difficulty placing outgoing calls due to inaccurate detection of network dial tone.
Valid Entry broadband medium
precise
Usage
This is the least exact of the levels of tone detection. If Avaya Communication Manager detects any tone at all, it interprets this as dial tone.
The server running Avaya Communication Manager interprets any tone which has a continuous "on" period of longer than 1 second as dial tone. Otherwise, the server accepts whatever the tone detector circuit pack reports.
Communication Manager accepts whatever the tone detector circuit pack reports.
Misoperation Alerting
Valid Entry y
n
Usage
Enables misoperation recall alerting on multi-appearance stations, analog stations, and attendant consoles.
Uses standard misoperation handling without recall alerting.
Network Feedback During Tone Detection
If enabled, provides audible feedback to the user while the system attempts to detect dial tone.
On-hook Dialing on 607/2400/4600/6400/8400 Terminals
If enabled, users can dial without lifting the handset. Users hear dial tone when they press the Speaker button, even if the handset is on-hook.
For 6400/8400, 607, 2420, 2410, and 4600 telephone users with speakerphones.
Outpulse Without Tone
Valid Entry y n
Usage The server outpulses digits even when a dial tone has not been received. The calling party receives intercept tone if no dial tone is detected.
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Feature-related system parameters: page 10
Pull Transfer
Enables or disables the Pull Transfer feature on a system-wide basis. With this, the transferring or transferred-to party can press the Transfer button to complete the transfer operation.
Repetitive Call Waiting Interval (sec)
Available only if the Repetitive Call Waiting Tone is enabled.
Valid Entry 1 to 99
Usage The number of seconds between call waiting tones.
Repetitive Call Waiting Tone
If enabled, a repetitive call waiting tone is provided to the called party for all types of call waiting access.
Send DTMF Over Telecommuter Link
In the telecommuter mode, when a user is on a call and presses any digits from a deskphone or a softphone, the user hears loud DTMF tones over the telecommuter link. Use the Send DTMF Over Telecommuter Link field to send or prevent DTMF tones over the telecommuter link.
Valid entry y
n
Usage
Communication Manager sends DTMF tones over the telecommuter link. The default value is y.
Communication Manager does not send DTMF tones over the telecommuter link.
Station Tone Forward Disconnect
Applies to any station other than one administered as a data endpoint, an attendant console, a BRI telephone, an auto answer, or as an Outgoing Call Management (OCM) agent.
Valid Entry busy intercept silence
Usage
The treatment given to a station that is the last party remaining off-hook on a call, until that station is placed on-hook, or until the tone has played for 45 seconds and is followed by silence.
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Feature-related system parameters
System Updates Time On Station Displays
If enabled, automatically updates the system time on two-line display telephones when you run background maintenance, for example, when you plug in the telephone set. The system sends the time update periodically to the telephones. This parameter does not apply to telephones such as the BRI telephones, where the user sets the time.
Update Transferred Ring Pattern
If enabled, changes the ringing pattern from internal to external when an internal station transfers an external call. Use this feature if most calls go through an attendant, so that users are able to distinguish an external call.
Vector Disconnect Timer (min)
Valid Entry 1 to 240
blank
Usage
Determines when the trunk gets disconnected if incoming or outgoing disconnect supervision is disabled for the trunk group.
Avaya Communication Manager does not initiate a disconnect.
Wait Answer Supervision Timer
Valid Entry y n
Usage Calls to stations unanswered after 50 seconds are dropped. Unanswered calls drop only when the calling party goes on-hook.
Related links Unanswered DID Call Timer (seconds) on page 310
ITALIAN DCS PROTOCOL
The next three fields control the Italian DCS Protocol feature.
Apply Intercept Locally
If enabled, DID/CO intercept treatment is applied locally instead of on the originating server or switch. Available only if Italian Protocol is enabled.
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Feature-related system parameters: page 11
Enforce PNT-to-PNT Restrictions
If enabled, restrictions and denial of PNT-to-PNT connections are enforced when the EDCS message is unavailable. Available only if Italian Protocol is enabled.
Italian Protocol Enabled
Enables or disables the Italian DCS feature on a system-wide basis.
Feature-related system parameters: page 11
CALL CENTER SYSTEM PARAMETERS EAS
Delay
Available only if Expert Agent Selection (EAS) or ASAI Link Core Capabilities are enabled for the system.
Valid Entry 0 to 99 blank
Usage
The number of seconds the caller hears ringback before the Direct Agent Announcement is heard by the calling party.
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Feature-related system parameters
Direct Agent Announcement Extension
The extension of the direct agent announcement.
Expert Agent Selection (EAS) Enabled
Enables or disables Expert Agent Selection (EAS). Enable only if ACD or vectoring hunt groups exist or if existing ACD or vectoring hunt groups are skill-based. Available only if EAS is enabled for the system.
Message Waiting Lamp Indicates Status For
Available only if Expert Agent Selection (EAS) is enabled for the system.
Valid Entry station
loginID
Usage
The message waiting lamp on a telephone indicates the message is for the telephone extension.
The message waiting lamp on a telephone indicates the message is for the agent login ID.
Minimum Agent-LoginID Password Length
The minimum number of digits that must be administered as an EAS Agent's LoginID password. Accepts up to nine digits. Available only if Expert Agent Selection (EAS) is enabled for the system.
VECTORING
Available Agent Adjustments for BSR
If activated, Best Service Routing (BSR) adjustments are made available to available agents. Available only if Vectoring (Best Service Routing) is enabled for the system.
BSR Tie Strategy
Available only if Vectoring (Best Service Routing) is enabled for the system.
Valid Entry 1st-found alternate
Usage
BSR uses the first selection for routing. This is the default.
Alternating the BSR selection algorithm is possible when a tie in Expected Wait Time (EWT) or available agent criteria occurs. Every other time a tie occurs for calls from the same active Vector Directory Number (VDN), the selection from the step with the tie is used instead of the first selected split/skill or location to send the call. This helps balance the routing over the considered local splits/skills and remote locations when cost of routing remotely is of least concern.
Converse First Data Delay
Available only if Vectoring (Basic) is enabled for the system.
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Feature-related system parameters: page 11
Valid Entry 0 to 9
Usage
The number of seconds data is prevented from being outpulsed as a result of a converse vector step from the system to Avaya IR before IR is ready. The delay commences when the IR port answers the call.
Converse Second Data Delay
Available only if Vectoring (Basic) is enabled for the system.
Valid Entry 0 to 9
Usage
Number of seconds used when two groups of digits are being outpulsed, as a result of a converse vector step, from the system to Avaya IR.
Prevents the second set from being outpulsed before IR is ready. The delay commences when the first group of digits has been outpulsed.
Converse Signaling Pause
The length in milliseconds of the delay between digits being passed. The optimum timer settings for Avaya IR are 60 msec tone and 60 msec pause.
Available only if Vectoring (Basic) and DTMF feedback signals are enabled for the system.
Valid Entry
40 to 2550 (in increments of 10)
Usage
Values are rounded up or down depending upon the type of circuit pack used to outpulse the digits.
· TN742B or later suffix analog board -- Rounds up or down to the nearest 25 msecs. For example, a 130 msec tone rounds down to 125 msecs, a 70 msec pause rounds up to 75 msec for a total of 200 msecs per tone
· TN464F, TN767E or later suffix DS1 boards -- Rounds up to the nearest 20 msecs. For example, a 130 msec tone rounds up to 140 msecs, a 70 msec pause rounds up to 80 msecs for a total of 220 msecs per tone.
If a circuit pack has been used for end-to-end signaling to IR, and has then been used to send digits to a different destination, IR timers might stay in effect. To reset the timers to the system default, pull and reseat the circuit pack.
Converse Signaling Tone
The length in milliseconds of the digit tone for digits being passed to Avaya IR. The optimum timer settings for IR are 60 msec tone and 60 msec pause.
Available only if Vectoring (Basic) and DTMF feedback signals are enabled for the system.
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Feature-related system parameters
Valid Entry
40 to 2550 (in increments of 10)
Usage
Values are rounded up or down depending upon the type of circuit pack used to outpulse the digits.
· TN742B or later suffix analog board -- Rounds up or down to the nearest 25 msecs. For example, a 130 msec tone rounds down to 125 msecs, a 70 msec pause rounds up to 75 msec for a total of 200 msecs per tone
· TN464F, TN767E or later suffix DS1 boards -- Rounds up to the nearest 20 msecs. For example, a 130 msec tone rounds up to 140 msecs, a 70 msec pause rounds up to 80 msecs for a total of 220 msecs per tone.
If a circuit pack has been used for end-to-end signaling to IR, and has then been used to send digits to a different destination, IR timers might stay in effect. To reset the timers to the system default, pull and reseat the circuit pack.
Interflow-qpos EWT Threshold
Available only if the Lookahead Interflow (LAI) is enabled for the system.
Valid Entry 0 to 9 blank
Usage
Number of seconds for this threshold. Any calls predicted to be answered before this threshold are not interflowed, therefore saving CPU resources.
Prompting Timeout (secs)
Available only if Vectoring (Prompting) is enabled for the system.
Valid Entry 4 to 10
Usage
The number of seconds before the Collect Digits command times out for callers using rotary dialing.
Reverse Star/Pound Digit for Collect Step
The "*" is interpreted as a caller end-of-dialing indicator and the "#" is an indicator to clear all digits previously entered by the caller for the current collect vector step.
Valid Entry y
n
Usage
Reverses the star and pound digits for the collect vector step. This does not affect any other vector step or other non-ACD feature, such as ARS.
The "*" and "#" digit-processing is unchanged.
Store VDN Name in Station's Local Call Log
Enables or disables the sending of a message from Communication Manager telling the telephone to store the VDN name or the calling party's name in the station call log for any of the following telephones:
· 2420
· 4610
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· 4620 · 4625
Feature-related system parameters: page 11
Service Observing field descriptions
Security alert: The use of Service Observing features might be subject to federal, state, or local laws, rules or regulations, or require the consent of one or both parties in the conversation. You must familiarize yourself and comply with all the applicable laws, rules, and regulations before using Service Observing features.
Field title
Field description
Allow Two Observers in Same Call
Controls whether two observers can monitor the same call. Valid entries are y and n (default).
If you select y, two observers can monitor the same Expert Agent Selection (EAS) agent login ID or station extension and up to two service observers can be on the same two-party call or in a conference call with more than two parties.
If you select n, only one service observer can monitor the EAS agent login ID or station extension.
Service Observing: Warning Controls whether the system plays a warning tone that the phone users
Tone
and calling parties can hear when a supervisor monitors calls.
Valid entries are y and n (default).
The option is applicable if the Service Observing (Basic) field on the System-Parameters Customer-Options screen is active for the system.
Service Observing: Conference Tone
Controls whether the system plays a conference tone that the phone users and calling parties can hear when a supervisor monitors calls.
Valid entries are y and n (default).
The option is applicable if the Service Observing (Basic) field on the System-Parameters Customer-Options screen is active for the system.
The option is not applicable if you set the Service Observing: Warning Tone option to y.
Table continues...
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Feature-related system parameters
Field title Allowed with Exclusion: Service Observing
Allowed with Exclusion: SSC
Coach on SSC
Field description
Controls whether multi-appearance phone users can prevent other service observers from bridging on to an existing call.
Valid entries are y and n (default).
If you select y, Communication Manager allows a Service Observing connection towards a station with Exclusion active, either by Class of Service (COS) or by manual activation of Exclusion.
If you select n, Communication Manager denies a Service Observing connection towards a station with Exclusion active. If an agent activates Exclusion while being observed, Communication Manager drops all bridged parties including the observer.
Controls whether multi-appearance phone users can prevent ASAI Single Step Conference (SSC) parties from bridging on to an existing call.
Valid entries are i, b, and n (default).
If you select i, Communication Manager allows invisible SSC parties a connection towards a station with Exclusion active, either by COS or by manual activation of Exclusion.
If you select b, Communication Manager allows both visible and invisible SSC parties a connection towards a station with Exclusion active, either by COS or by manual activation of Exclusion.
If you select n, Communication Manager denies a SSC connection towards a station with Exclusion active. If an agent activates Exclusion while being observed, Communication Manager drops all bridged parties including the observer.
Controls whether a supervisor can provide whisper coaching on a call with an invisible SSC. When using whisper coaching, only the agent will hear the supervisor, not any other parties on the call.
Valid entries are y and n (default).
If you select y, the supervisor can change to the coaching mode to give advice to the agent without the customer hearing that advice. The supervisor is heard by the agent, but other parties on the call will not hear the supervisor. If you are also using Avaya Call Recording, anything said by the supervisor is recorded except while in the coaching mode.
If you select n, the supervisor is blocked from changing to the coaching mode while the invisible SSC is on the call.
Note:
When using the "sip-sobsrv" feature button, coaching is only available from Call Center Elite SIP devices.
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Feature-related system parameters: page 12
Feature-related system parameters: page 12
AGENT AND CALL SELECTION
ACW Agents Considered Idle
Valid Entry y
n
Usage
Include agents who are in After Call Work (ACW) mode in the Most-Idle Agent queue. This means that ACW is counted as idle time.
Exclude ACW agents from the queue.
Auto Reserve Agents
When a critical skill is not meeting its service level, auto-reserve puts agents in standby for their other skills to ensure that there is an available agent when the next call arrives for the critical skill. When an agent becomes available, all of his or her assigned skills are checked to see if any autoreserve skills are not meeting their target service level. If so, the agent is made available only in those skills.
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Feature-related system parameters
Valid Entry all none secondary-only
Usage Puts an agent on standby for all skills. Agent is not on standby for any additional skills. Puts an agent on standby only for secondary skills.
Block Hang-up by Logged-In Auto Answer Agents
The system displays the field only if you set the EAS field to y. The enhancement is available for agents using H.323 IP and DCP telephones, but does not apply and is not required for agents using BRI, analog, CallMaster, or SIPCC telephones.
Valid Entry n
y
Usage If an auto-answer agent goes on-hook when on an ACD call:
· Communication Manager drops the active call and logs the agent out.
· The reporting adjunct fails to record the call and track the logged-in agent. You, as an administrator, can prevent an agent in the auto-answer mode from accidently logging out or dropping an active call.
The agent can perform the following functions to drop calls:
· Use the release button to disconnect the call and stay logged-in.
· Use the drop button to release the last connected party.
Note:
If you set the field to y and the logged-in agent appearance is on-hook, a new call is not delivered automatically. The new call rings at the station and the agent must manually answer the call. If you have administered RONA for the skill, the call is redirected if the agent fails to answer within the administered number of rings.
Call Selection Measurement
Determines how Avaya Communication Manager selects a call for an agent when the agent becomes available and there are calls in queue.
Valid Entry current-wait-time predicted-wait-time
Usage
Selects the oldest call waiting for any of the agent's skills.
Uses the time a call is predicted to wait in queue instead of the time the call has already waited. Available only if Business Advocate features are enabled for the system.
Copy ASAI UUI During Conference/Transfer
Enables or disables copying user-to-user (UUI) information during a conference call or during call transfers. Available only if ASAI Link Core Capabilities or Computer Telephony Adjunct Links are enabled for the system.
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Note:
The system copies all UUI information, not just ASAI UUI. Copying occurs only during a human-initiated conference or transfer. Communication Manager does not copy the UUI if the conference or transfer is initiated by ASAI.
MIA Across Splits or Skills
Enables or disables the removal of an agent from the Most Idle Agent (MIA) queue for all available splits/skills/hunt groups when answering a call from any of these groups.
Service Level Maximizer Algorithm
Selects an alternative algorithm for selecting agents and delivering calls to maximize service level targets. Available only if Service Level Maximizer is enabled for the system.
Valid Entry actual
weighted
Usage
The Actual Service Level (ASL) is determined as a percentage on a hunt group basis using the number of accepted calls in the current interval divided by the total calls in the current interval. A call is counted as accepted if it is answered within the target service level time period.
The Weighted Service Level (WSL) is based on a weighting calculation that uses the difference between the target time and the estimated wait time.
Service Level Supervisor Call Selection Override
Determines whether Avaya Communication Manager changes agent call handling preferences when a skill using Service Level Supervisor exceeds its Level 1 threshold. Available only if Expert Agent Selection (EAS) and the Business Advocate features are enabled for the system.
Valid Entry y
n
Usage
Overrides the normal call handling preferences of a skill's assigned agents in this situation.
Normal call handling preferences are in effect when the skill exceeds its Level 1 threshold.
ASAI
Call Classification After Answer Supervision?
For use with ASAI Outbound Call Management (OCM).
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Feature-related system parameters
Valid Entry y
n
Usage
Forces the server running Communication Manager to rely on the network to provide answer/busy/drop classification to the server. After the call has been answered, a call classifier can be added to perform answering machine, modem, and voice answering detection.
Always connects a classifier after call setup for determining call progress and answer. ISDN progress messages generally take precedence.
Send UCID to ASAI
Enables or disables the transmission of Universal Call ID (UCID) information to ASAI.
CALL MANAGEMENT SYSTEMS
REPORTING ADJUNCT RELEASE (determines protocol used by appl link)
CMS (appl mis)
Use this field to specify the release of the Call Management System (CMS) reporting adjunct. Management Information System (MIS) is a communication-interface link that connects CMS to Communication Manager.
Valid entry R12
R13
R13.1
R14.1 R15/R16 R16.1/R16.x R19.1+ (secured) blank
Usage CMS Release 12 connects to Communication Manager through the mis1 link.
CMS Release 12 is incompatible with Avaya IQ. CMS Release 13 connects to Communication Manager through the mis1 link.
CMS Release 13 is incompatible with Avaya IQ. CMS Release 13.1 connects to Communication Manager through the mis1 link.
CMS Release 13.1 and later is compatible with Avaya IQ. CMS Release 14.1 connects to Communication Manager through the mis1 link. CMS Release 15 and 16 connects to Communication Manager through the mis1 link. CMS Release 16.x and 17 connects to Communication Manager through the mis1 link. CMS Release 19.1 or higher connects to Communication Manager through the TLS link. CMS does not connect to Communication Manager. This is the default value.
In the case of two CMS adjuncts, one CMS adjunct connects to Communication Manager through the mis1 link and the other CMS adjunct connects to Communication Manager through the mis2 link.
For more information about CMS (appl mis), see Administering Avaya Aura® Call Center Elite.
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IQ (appl ccr)
Expert Agent Selection (EAS) and Universal Call ID (UCID) must be enabled before a connection can be established with Avaya IQ.
Valid Entry 4.0 5.0 5.1 blank
Usage The release of Avaya IQ connected to the ccr1 link, and to the ccr2 link for a second Avaya IQ.
Avaya IQ is disconnected. This is the default.
OTHER CALL MANAGEMENT SYSTEM FIELDS
ACD Login Identification Length
Available only if Expert Agent Selection (EAS) is disabled and BCMS/VuStats Login IDs are enabled for the system.
Valid Entry 1 to 9
Usage
The number of digits for an ACD agent login ID that identifies an ACD agent to the Call Management System. This number must equal the number of characters in the agent's login ID and cannot be 0.
BCMS/VuStats Abandon Call Timer (seconds)
Valid Entry 1 to 10 blank
Usage
The number of seconds before calls are abandoned. Calls with talk time that is less than this number and are not held, are tracked by the Basic Call Management System (BCMS) and displayed by VuStats as ABAND calls.
BCMS/VuStats LoginIDs
Enables or disables valid agent login IDs to monitor call activity by an agent. Basic Call Management System (BCMS) and VuStats login IDs are available in addition to Expert Agent Select (EAS) login IDs if EAS is enabled on the system. Both BCMS and CMS use the same login ID for an agent.
BCMS/VuStats Measurement Interval
Available only if BCMS (Basic) or VuStats is enabled for the system.
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Feature-related system parameters
Valid Entry half-hour hour
Usage
Selects measurement intervals for polling and reporting data. There are a maximum of 25 time slots available for measurement intervals. If hour is specified, an entire day of traffic information will be available for history reports. Otherwise, only half a day will be available. This does not affect daily summaries as they always reflect traffic information for the entire day. The interval can be changed at any time, but remains until the current interval completes.
Clear VuStats Shift Data
Valid Entry on-login at-midnight
Usage Clears shift data for an agent when the agent logs in. Clears shift data for all agents at midnight.
Remove Inactive BCMS/VuStats Agents
Valid Entry y
n
Usage
Agents are removed from reports when they have no staff time during the previous seven days.
Agents remain on the report even if they have no staff time for any period of time.
Validate BCMS/VuStats Login IDs
Valid Entry y
n
Usage
Only login-IDs that have been administered for the Basic Call Management System (BCMS) can be entered.
Any ACD login of the proper length can be entered.
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Feature-related system parameters: page 13
Feature-related system parameters: page 13
CALL CENTER MISCELLANEOUS
Allow Ringer-off with Auto-Answer
If activated, an agent can use the ringer-off feature button that prevents ringing on Expert Agent Selection (EAS) auto-answer calls.
Callr-info Display Timer (sec)
Valid Entries 3 to 60
Usage
Administer the timer to display the information which caller has dialed on IP (H.323) telephones. The default value is 10 secs.
Clear Callr-info
Specifies when the collected digits Callr-Info display is removed from the agent or station display.
Valid Entry leave-ACW next-call
Usage Leaves the display up while the agent is in After Call Work (ACW) mode. Clears the display when the next call is received. This is the default.
Table continues...
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Feature-related system parameters
Valid Entry on-call-release
Usage
Clears the display on the second line of a two-line display telephone or the display of a one-line display telephone with the Enhanced Callr-Info display for 1-line display phones option active when the call is released. The call is released if the receiving call is disconnect or if the agent presses the release button.
Interruptible Aux Notification Timer (sec)
Valid Entry 1 to 9
Usage
Specifies the number of seconds the endpoint interruptible aux notifications, the flashing lamp, display, or tone, are on before an auto-in-interrupt or manual-ininterrupt agent is made available. This delay ensures that an agent is not immediately made available when he presses an interruptible aux button. Also, this delay provides a brief period to an agent already in interruptible Aux mode before that agent is made available automatically. Default is 3.
PC Non-Predictive Reports Skill
Administers a skill hunt group used for reporting associated with Proactive Contact non-predictive switch-classified calls on a per-system basis. Reports are generated as though the agent were in the ACD-OUT state. Available only if reporting for PC Non-Predictive Calls is enabled.
Valid Entry 1 to 99 1 to 8000
Usage Skill number for S8300E switches. Skill number for all other platforms
Prefer H.323 Over SIP For Dual-Reg Station 3PCC Make Call
The Prefer H.323 Over SIP For Dual-Reg Station 3PCC Make Call field is used to enable the Redirect 3PCC to H.323 station from SIP desktop station feature.
For more information about the Redirect 3PCC to H.323 station from SIP desktop station feature, see Avaya Aura® Communication Manager Feature Description and Implementation.
Valid entries y
n
Usage
Redirects the 3PCC actions from a SIP desktop station to a remote soft phone or virtual private network H.323 telephone.
Does not redirect the 3PCC actions from a SIP desktop station to a remote soft phone or virtual private network H.323 telephone.
Reporting for PC Non-Predictive Calls
Activates or deactivates improved integration with Proactive Contact Outbound Calling for non switch-classified outbound calling. For example, this feature improves Call Management System tracking for switch-classified and non-switch classified (agent classified) outbound calls placed by the Proactive Contact soft dialer through ASAI. Default is n.
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Send Connect Event to ASAI For Announcement Answer
Use this field to enable Communication Manager to send a Connected or Answered event to the calling party if the calling party is an ASAI application and the called party uses an announcement system. If you set this field to y, Communication Manager sends the Connected Answered event to the ASAI application when the announcement system at the called party answers the call. The announcement system answers the call by collecting the digits that the ASAI application provides. The system uses the collect command of the vector to collect the digits.
Note: Communication Manager does not send a Drop event when the call gets disconnected.
Valid entry n
y
Usage
Communication Manager does not send the Connected or Answered event to the ASAI application. This is the default value.
Communication Manager sends the Connected or Answered event to the ASAI application.
Note:
You should be careful while creating vectors that include announcements and other advanced features like route requests. The features can lead to:
· Call sequences in which Communication Manager sends a Connected event to the called party but do not send an Alerting event.
· Call sequences in which Communication Manager sends a Connected event to a calling party followed by an Alerting event to the same party but with different domains associated with the Connected event and the Alerting event.
Service Level Algorithm for SLM
Administers the algorithm used to determine service levels for Service Level Maximizer (SLM). Available only if Expert Agent Selection (EAS) has been enabled for the system.
Valid Entry actual
weighted
Usage
The Actual Service Level (ASL) based on an algorithm that is determined as a percentage on a hunt group (skill) basis using the number of calls answered within the target service level time period in the current interval, divided by the total calls in the interval. This is the default.
The Weighted Service Level (WSL) algorithm based on a weighting calculation that uses the difference between the target time and the estimated wait time.
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Feature-related system parameters
Zip Tone Burst For Callmaster Endpoints
Valid entry double
single
Usage
When the station set type is a Callmaster series, Communication Manager retains existing operation which applies the following to ACD agents:
· two bursts of zip tone for auto-answer ACD calls
· two burst of ICI tweedle-dee tone for non-ACD auto-answer calls
This is the default value.
Communication Manager eliminates the 2nd burst of zip/ICI tone reducing time for agent to start conversation with the caller and possibility of the agent and the caller hear open mike background noise between the first and second tones. Use with a Callmaster station type when the agent can always hear enough of the single burst auto-answer to recognize that a call is being delivered.
Feature-related system parameters: page 14
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Feature-related system parameters: page 14
REASON CODES
Aux Work Reason Code Type
Valid Entry none requested
forced
Usage
An agent does not enter a Reason Code when entering AUX work.
An agent can enter a Reason Code when entering AUX mode but is not forced to do so. Available only if Reason Codes and EAS are enabled for the system.
An agent is forced to enter a Reason Code when entering AUX mode. Available only if Reason Codes and EAS are enabled for the system.
Logout Reason Code Type
Valid Entry none requested
forced
Usage
An agent does not enter a Reason Code when logging out.
An agent can enter a Reason Code when logging out but do not want to force the agent to do so. Available only if Reason Codes and EAS are enabled for the system.
An agent is forced to enter a Reason Code when logging out. Available only if Reason Codes and EAS are enabled for the system.
Redirection on No Answer Aux Work Reason Code
Redirection on No Answer (RONA) redirects a ringing ACD skill or direct agent call after an administered number of rings. When RONA is invoked for a call, the system places an agent in the Aux work mode, and the agent is unavailable to receive calls from other splits and skills. This field is used to define the Reason Code for reporting, instead of the default of 0, when the state of the agent is changed to Aux due to RONA.
Valid Entry 0 to 99 0 to 9
Usage If the Two-Digit Aux Work Reason Code field is set to y. If the Two Digit Aux Work Reason Code field is set to n.
ROOF Failure and Unreachable Aux Work Reason Code
This field provides redirection of calls back into queue or to the specified VDN. The agent is not aware that the line is in the Aux work mode during an IP connectivity failure. If connectivity is restored during the IP network TCP retry period, the lamp will indicate that the line is in the Aux work mode. The value in the field is used to define the Reason Code for reporting, instead of the default of 0, when the state of the agent is changed to Aux due to ROOF.
This field applies only to ACD agents using SIP hard or soft phones administered as Off-PBX Telephone Integration and Mobility (OPTIM) endpoints
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Feature-related system parameters
Valid entry 0 to 99 0 to 9
Usage If the Two-Digit Aux Work Reason Code field is set to y. If the Two Digit Aux Work Reason Code field is set to n.
Two-Digit Aux Work Reason Codes
Enables or disables two-digit reason codes for agent state changes for AUX Work.
REDIRECTION ON IP CONNECTIVITY FAILURE
Auto-answer IP Failure AUX Reason Code
Valid Entry 0 to 99
Usage
The reason code assigned for auto-answer IP failure, as the reason the agent was put into AUX Work.
Switch Hook Query Response Timeout
Valid Entry 500 to 5000 (msec)
blank
Usage
The time that call processing waits for a response from the switch hook query before Return on IP Connectivity Failure (ROIF) is triggered.
ROIF is inactive.
MAXIMUM AGENT OCCUPANCY PARAMETERS
The Maximum Agent Occupancy (MAO) threshold is a system-administered value that is applied across all administered agents and is based on the total percentage of agent time in call service. MAO data is derived from the same calculations that are used to derive Least Occupied Agent (LOA).
When an agent who exceeds the specified MAO threshold attempts to become available, he or she is automatically placed in AUX mode for the reason code administered for this purpose. When the occupancy for such pending agents drops below the MAO, they are released from AUX mode and made available.
Maximum Agent Occupancy AUX Reason Code
Valid Entry 0 to 99
Usage A reason code value. Do not use reason code 0. Default is 9.
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Maximum Agent Occupancy Percentage
Valid Entry 0 to 100
Usage
The maximum percentage of time an agent can be taking calls. This time is based on Maximum Agent Occupancy (MAO) calculations. Default is 100.
Feature-related system parameters: page 15
FORCED AGENT LOGOUT PARAMETERS
ACW Forced Logout Reason Code
Available only if the Call Center Release is 3.0 or later and Expert Agent Selection (EAS) is enabled for the system.
Valid Entry 0 to 9
Usage The reason for logging out the agent due to time-out in After Call Work (ACW).
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Feature-related system parameters
Clock Time Forced Logout Reason Code
Valid Entry 0 to 9
Usage
The reason code for the Forced Agent Logout by Clock Time feature used by administrators to set a specific time when the system automatically logs out Expert Agent Selection (EAS) agents.
Maximum Time Agent in ACW before Logout (sec.)
Available only for Call Center Release 3.0 or later and if Expert Agent Selection (EAS) is enabled for the system.
Valid Entry 30 to 9999
blank
Usage
A system-wide maximum time an agent can be in After Call Work (ACW). When this timer expires, the agent is logged out.
There is no time-out. This is the default.
Forced Agent Aux Work by Location Reason Code
This field to displays the reason that all agents in a location are put in the AUX work mode. The valid entries are from 0 to 9. The default value is 0.
Forced Agent Aux Work by Skill Reason Code
This field to displays the reason that all agents of a skill put in the AUX work mode. The valid entries are from 0 to 9. The default value is 0.
Forced Agent Logout by Location Reason Code
This field to displays the reason that all agents in a location are logged out. The valid entries are from 0 to 9. The default value is 0.
Forced Agent Logout by Skill Reason Code
This field to displays the reason that all agents of a skill are logged out. The valid entries are from 0 to 9. The default value is 0.
Forced SIP Agent Logout for Unreachable Reason Code
The field indicates the reason that an agent is logged out due to the final unreachable failure. You can set a value from 0 to 9. By default, the value is set to 0.
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Feature-related system parameters: page 15
SIP STATION REACHABILITY CHECKING OPTIONS
Enable Reachability for Domain Control SIP Stations
Using the Enable Reachability for Domain Control SIP Stations-y/n/disable-all field, you can enable or disable options polling for specific stations. The options are:
Valid entry Y
N
disable-all
Usage
To enable the polling feature for an individual SIP station.
To disable the polling feature for an individual SIP station.
To apply systemwide values.
Enable SIP Agent Reachability?
Use the field to enable the Call Center Elite agents to enable or disable Options polling.
SIP Reachability Polling Interval
The field defines the frequency of OPTIONS message sent by Communication Manager. The field has a range from 5 to 30 minutes. By default, the field is set to 5 minutes.
SIP Station Reachability Attempts
The field administers the number of times an Options poll fails before taking the final action with the polling during error response polling. This field has a range from 1 to 5, and 3 is set by default.
SIP Unreachable Polling Period
The field defines the period of time that the OPTIONS messaging continues after Communication Manager polls and declares the agent as unreachable. The field has a range from 0 to 1440 minutes. By default, the field is set to 60 minutes.
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Feature-related system parameters
Feature-related system parameters: page 16
SPECIAL TONE
Special Dial Tone
Enables or disables an audible tone indicating that the station is locked. Related links
Special Dial Tone on page 309
Special Dial Tone for Digital/IP Stations
Valid Entry all none non-display
Usage
System plays special dial tone for digital or IP stations, if the Limit Number of Concurrent Calls, Station Lock, or Call Forward feature is active.
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Busy/Reorder/Intercept Tone Timeout (Seconds)
Valid Entry 3 to 45
Usage Allows you to configure the time to play busy, reorder, or intercept tone. Communication Manager releases the call appearance after the specified time.
This field is not applicable for SIP stations. This Timer is applicable only when Treatment type is tone. For other Treatment types like announcement, the timer will be ignored.
This field is applicable to the following fields:
· Controlled Outward Restriction Intercept Treatment
· Controlled Termination Restriction (Do Not Disturb)
· Controlled Station to Station Restriction
· DID/Tie/ISDN/SIP Intercept Treatment
· Invalid Number Dialed Intercept Treatment
· Invalid Number Dialed Display
· Restricted Number Dialed Intercept Treatment
· Restricted Number Dialed Display
REDIRECTION NOTIFICATION
Chained Call Forwarding
Enables or disables Chained Call Forwarding. With Chained Call Forwarding, calls can be forwarded to as many as 10 forwarded-to stations using a preset forwarding path.
Display Notification Enhanced Call Forward
Enables or disables display notification for Enhanced Call Forward.
Display Notification for a locked Station
Enables or disables display notification for a locked station.
Display Notification for Call Forward
Enables or disables display notification for Call Forward.
Display Notification for Do Not Disturb
Enables or disables display notification for Do Not Disturb.
Display Notification for Limit Number of Concurrent Calls
Enables or disables display notification for Limit Number of Concurrent Calls.
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Feature-related system parameters
Display Notification for Posted Messages
Enables or disables display notification for posted messages.
Display Notification for Send All Calls
Enables or disables display notification for Send All Calls.
Scroll Status messages Timer (sec.)
Valid Entry 5 to 10
blank
Usage The time in seconds for displaying scroll status messages. The status information shows the feature that has the highest priority and is enabled. The features in order of decreasing priority are:
· Do Not Disturb
· Send All Calls
· Call Forward
· Posted Messages
· Limit Number of Concurrent Calls (LNCC)
· Station Lock Deactivates the scrolling. This is the default.
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Feature-related system parameters: page 17
AUTOMATIC EXCLUSION PARAMETERS
Automatic Exclusion by COS
Valid Entry y
n
Usage
Enables automatic exclusion by a class of service when a user goes off-hook on a station with an assigned Exclusion button. This works only for stations on the local server running Communication Manager.
Exclusion operates normally.
Automatic Exclusion Coverage/Hold
Available only when Automatic Exclusion by COS is enabled. This field is applied only for "DND/SAC/Goto Cover" in Coverage path. This field is not applied for Bridge Call Appearance.
Valid Entry y
n
Usage
The principal can bridge onto the call by pressing the appropriate bridged appearance button. And, if the coverage point places the exclusion call on hold, the principal can retrieve the call.
If a coverage point has answered a call and there is active exclusion on the call, the principal cannot bridge onto the call. And, if the coverage point places the exclusion call on hold, the principal cannot retrieve the call.
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Feature-related system parameters
Automatic Exclusion with Whisper Page
Available only Automatic Exclusion by COS is enabled.
Valid Entry y n
Usage
The whisper page goes through to an excluded call.
The whisper page is denied when a station attempts to whisper page to a station that is on an excluded call.
Duration of Call Timer Display
Valid Entry 3 to 30
Usage
The length of time in 3 second increments that the call information remains on display after the call is terminated.
Password to Change COR by FAC
If this field contains a value, a password option is required. Accepts from four to eight digits. Available only if Change COR by FAC is enabled for the system.
Recall Rotary Digit
Valid Entry 0 to 9
Usage
The digit used for rotary telephones to receive recall dial tone. Dialing this digit simulates switch-hook flash so that users of rotary telephones can use features such as conference and transfer. The telephone must also be administered to use the recall rotary. This should be a number that is not the first digit in normal dialing patterns.
WIRELESS PARAMETERS
Radio Controllers with Download Server Permission
The port location of the circuit pack containing the radio controllers with download server permission.
Valid Entry 1 to 64 A to E 0 to 20
Usage First and second characters are the cabinet number Third character is the carrier. Fourth and fifth characters are the slot number.
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Feature-related system parameters: page 17
IP PARAMETERS
Direct IP-IP Audio Connections
Use this field to enable direct audio connections between IP endpoints. Direct audio connections save bandwidth resources and improve the sound quality of voice over IP transmissions.
Note: This field has no impact on a SIP station (OPS) on shuffling. For SIP stations, this field is controlled by the "Direct IP-IP Audio Connections" field in the Signaling group form.
IP Audio Hairpinning
If enabled, IP endpoints connect through the IP circuit pack in the server in IP format to bypass the Communication Manager TDM bus.
RUSSIAN MULTI-FREQUENCY PACKET SIGNALING
Re-try
Enables or disables the resending of address information on outgoing Russian MFP trunks. Specifically, the server running Communication Manager resends Russian MFP calling party number and dialed number information to the local telephone company central office (CO). The server resends the information only once over another outgoing trunk port of the same trunk group if Communication Manager receives a message that the information was received incorrectly by the CO. The switch also sends Russian MFP information over another trunk port if Communication Manager does not receive a timely response for the information.
T2 (Backward Signal) Activation Timer (secs)
Valid Entry 5 to 20
Usage
The number of seconds Communication Manager waits to receive confirmation after sending calling party number and dialed number information on outgoing Russian MFP trunks.
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Feature-related system parameters
Feature-related system parameters: page 18
INTERCEPT TREATMENT PARAMETERS
Intercept Treatment on Failed Trunk Transfers
Valid Entry y n
Usage Provides intercept treatment to calls failing trunk transfers. Drops calls failing trunk transfers.
Invalid Number Dialed Display
Used to display a name in either Latin or Asian characters for an invalid number calling in. This field supports both a NAME1 and a NAME2 value. A NAME1 value directs the system to use the table of names that contains Latin characters that can be displayed. A value of NAME2 directs the system to use the UTF-8 table of names that contains non-ASCII characters suitable for Asian language names. Accepts up to 15 alphanumeric characters.
Note:
The system displays the message administered for the Invalid Number Dialed Display field on the telephone only if the Invalid Number Dialed Intercept Treatment field is set to announcement.
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Feature-related system parameters: page 18
Invalid Number Dialed Intercept Treatment
The type of intercept treatment the end-user hears after dialing an invalid number.
Valid Entry announcement
tone
Usage
Provides a recorded announcement when the end-user dials an invalid number. You select and record the message.
Requires entering the extension number for the announcement.
Provides intercept tone when the end-user dials an invalid number. This is the default.
Restricted Number Dialed Display
The string of alphanumeric characters assigned for calls that are denied because of COS, COR, or FRL restrictions. This field supports both a NAME1 and a NAME2 value. A NAME1 value directs the system to use the table of names that contains Latin characters. A value of NAME2 directs the system to use the UTF-8 table of names that contains non-ASCII characters suitable for Asian language names. Accepts up to 15 alphanumeric characters.
Note:
The system displays the message administered for the Restricted Number Dialed Display field on the telephone only if the Restricted Number Dialed Intercept Treatment field is set to announcement.
Restricted Number Dialed Intercept Treatment
The type of intercept treatment the caller hears after dialing a number restricted from them due to COS, COR, or FRL restrictions.
Valid Entry tone announcement
Usage
Provides intercept tone. This is the default.
Provides a recorded announcement. You select and record the message. Requires entering the extension number for the announcement.
WHISPER PAGE
Whisper Page Tone Given To
Determines who should hear a Whisper Page beep tone.
Valid Entry all paged
Usage All parties hear the whisper page beep tone. The whisper page feature sends a beep tone only to the paging and the paged party.
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Feature-related system parameters
6400/8400/2420J LINE APPEARANCE LED SETTINGS
Warning:
The following fields change only the LED operation for 84xx and 64xx model telephones. When the LED operation is changed using any of these fields, then IP Agent and IP Softphone using a station type of 84xx or 64xx does not work. For station types other than 84xx or 64xx, a change to the LEDs using these fields does not affect either IP Agent or IP Softphone.
Note:
The system generates a warning if the default values of the LED Settings field are changed. The warning message states WARNING: Avaya Softphone will not operate correctly if this value is changed This warning message displays for Avaya Communication Manager Release 3.1 or later.
Display Information With Bridged Call
Controls whether or not name and number for a bridged call are displayed on the telephone of the called party. This field does not control the content of the display.
Idle
Note:
This field applies only to 8400 and 6400 series telephones. The 2400 series telephone uses icons rather then LEDs.
Valid Entry steady off
Usage The LED flash rate for an idle station. Default is steady. The correct value for the Japanese environment is off.
Other Stations When Call Is Active
Controls a DCP bridged appearance LED for those non-active parties with a bridged appearance that is active.
Note:
This field applies only to 8400 and 6400 series telephones. The 2400 series telephone uses icons rather then LEDs.
Valid Entry green red
Usage The LED color. Default is green. Red is the correct value in the Japanese environment.
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Feature-related system parameters: page 18
Other Stations When Call Is Put On-Hold
Controls LED options for the other stations with a Bridged Appearance that have been placed on hold, but the user of this station has not pressed the hold button.
Note:
This field is for a DCP bridged appearance LED color and flash rate when a call on a bridged appearance is put on hold by another party on the DCP bridged appearance. Additionally, this field only applies to 8400 and 6400 series telephones. The 2400 series telephone uses icons rather then LEDs. Correct operation in the Japanese environment requires the administrator to select the values red and flash for this field.
Valid Entry green red off wink inverse-wink flash flutter broken-flutter steady
Usage The color of the LED. Default is green.
The flash rate for the LED. Default is wink.
Pickup on Transfer
Valid Entry y n
Usage
A station with a bridged appearance can pick up a call on hold because of a transfer.
Another station with bridged appearances cannot pick up a call on hold because of a transfer.
Ringing
Controls the LED color and flash rate while a call is ringing.
Note:
This field only applies to 8400 and 6400 series telephones. The 2400 series telephone uses icons rather then LEDs. Correct operation in the Japanese environment requires the administrator to select the values red and wink for this field.
Valid Entry green red
Usage The LED color. Default is green.
Table continues...
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Feature-related system parameters
Valid Entry off wink inverse-wink flash flutter broken-flutter steady
Usage The flash rate. Default is flash.
Station Putting Call On-Hold
Controls the LED color and flash rate on the 8400 and 6400 series telephones for a call held on a Primary or Bridged Appearance.
Valid Entry green red off wink inverse-wink flash flutter broken-flutter steady
Usage The color of the LED. The LED for the color not selected is turned OFF. Default is green.
The flash rate for a call on hold. Default is wink.
Station When Call is Active
Controls the red LED on the 8400 and 6400 series telephones, for a station active on a call.
Valid Entry steady off
Usage Communication Manager controls the red LED. This is the default. The red LED is always OFF.
Feature-related system parameters: page 19
change system-parameters features
Page 19 of 19
FEATURE-RELATED SYSTEM PARAMETERS
IP PARAMETERS
Direct IP-IP Audio Connections? y
IP Audio Hairpinning? n
Synchronization over IP? n
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Feature-related system parameters: page 19
SIP Endpoint Managed Transfer? n
Expand ISDN Numbers to International for 1XCES? n
CALL PICKUP
Maximum Number of Digits for Directed Group Call Pickup: 4
Call Pickup on Intercom Calls? y
Call Pickup Alerting? y
Temporary Bridged Appearance on Call Pickup? y
Directed Call Pickup? n
Extended Group Call Pickup: none
Enhanced Call Pickup Alerting? n
Call Pickup for Call to Coverage Answer Group? n Display Information With Bridged Call? n
Keep Bridged Information on Multiline Displays During Calls? y PIN Checking for Private Calls? n
IP PARAMETERS
Allow Bridge DM Answer
Use this field to enable the Bridge call answer as Direct Media feature. To enable this feature, set the SIP Endpoint Managed Transfer and Allow Bridge DM Answer fields to y.
Valid entry y
n
Usage
The calling party can have direct media path with the bridge call appearance of the called party station if all the involved stations support the Enhanced SIP Signaling feature.
This is the default value.
Direct IP-IP Audio Connections
Use this field to enable direct audio connections between IP endpoints. Direct audio connections save bandwidth resources and improve the sound quality of voice over IP transmissions.
Note:
This field has no impact on a SIP station (OPS) on shuffling. For SIP stations, this field is controlled by the "Direct IP-IP Audio Connections" field in the Signaling group form.
IP Audio Hairpinning
If enabled, IP endpoints connect through the IP circuit pack in the server in IP format to bypass the Communication Manager TDM bus.
Synchronization over IP
Enables or disables the Clock Synchronization over IP (CSoIP) feature to provide IP sync to a group of members.
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Feature-related system parameters
Valid entry y n
Usage Enables the CSoIP feature. Disables the CSoIP feature. This is the default value.
SDP Capability Negotiation for SRTP
The system displays SDP Capability Negotiation for SRTP when the Media Encryption Over IP field is set to y. Enables or disables SIP SRTP enhancements with Direct Media support in Communication Manager using Session Description Protocol (SDP) capability negotiation.
SIP Endpoint Managed Transfer
Use this field to enable the SIP Endpoint Managed Transfer feature. Before enabling the feature, you must ensure that the Network Call Redirection field is disabled for the SIP trunk group between Communication Manager and Session Manager.
Note:
The SIP Endpoint Managed Transfer field only controls the SIP transfer changes made in release 6.2. The field does not control some related Communication Manager SIP conference changes.
For more information on the SIP Endpoint Managed Transfer feature and on the related SIP conference changes, see Avaya Aura® Communication Manager Feature Description and Implementation.
Expand ISDN Numbers to International for 1XCES
Enable this field to convert ISDN calling numbers into the full international format when notifying Avaya one-X® Client Enablement Services about a call. This conversion is necessary in countries with complex numbering plans so that users can return calls from their one-X Mobile call log.
Valid entry y
n
Usage
Convert ISDN calling numbers into the full international format when notifying an Avaya one-X® Client Enablement Services server about a call.
If you enable this field, you must also configure the following:
· The Local E.164 Country Code field on the Location Parameters form: To convert a number from national to the international format.
· The City/Area column on the Locations form: To convert a number from local or subscriber format to the international format.
Do not convert the ISDN calling numbers to the full international format, when notifying an Avaya oneX® Client Enablement Services server about a call.
This is the default value.
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Feature-related system parameters: page 19
CALL PICKUP
A pickup group is a collection, or list, of individual extensions that have permission to answer calls originally dialed towards a different member of the group.
Call Pickup Alerting
Enables or disables Call Pickup Alerting on a system-wide basis. Call Pickup Alerting provides pickup group members with a visual indication on the Call Pickup status lamp for calls eligible to be answered using Call Pickup.
Call Pickup for call to Coverage Answer Group
Use this field to enable or disable interaction between Coverage Answer Group and Call Pickup group. Same extension can be configured to be part of coverage answer-group and pickup group.
Valid entries y
n
Usage
When a call rings at a Coverage Answer Group member, Communication Manager triggers call pickup alerting if the Coverage Answer Group member is part of a Call Pickup group.
When a call rings at a Coverage Answer Group member, Communication Manager does not trigger call pickup alerting if the Coverage Answer Group member is part of a Call Pickup group.
This is applicable for the H.323, DCP, Analog and SIP endpoints. The default value is n.
Call Pickup on Intercom Calls
System-wide users can attend or drop an intercom call using the Call Pickup or Directed Call Pickup features.
Directed Call Pickup
Activates or deactivates the use of Directed Call Pickup, where users can specify what ringing telephone they want to answer.
Enhanced Call Pickup Alerting
Valid entries y n
Usage Apply the Enhanced Call Pickup Delay Timer. Silently flash the pickup button.
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Feature-related system parameters
Audible Notification
Valid Entry 1 to 15 seconds
Usage
Set the audio notification timer. When the timer is set and a call arrives to a group member, the other members of the group are informed with an audio notification when the timer expires.
Enhanced Call Pickup Delay Timer (sec.) Display
Valid Entry 1 to 15
Usage
When a call arrives at one of the group members the other members are informed with a display notification. Set the timer in seconds to delay the display notification. When the timer expires the display notification is sent.
Extended Group Call Pickup
Selects how call pickup groups can answer calls directed to another call pickup group.
Valid Entry flexible simple none
Usage A one-to-n pickup group-to-extended pickup group mapping. A one-to-one pickup group-to-extended pickup group mapping. Extended group call pickup is prohibited.
Keep Bridged Information on Multiline Displays During Calls
If this field is enabled, for Multiline Display sets (1603, 1608, 1616, 4610, 4620, 4620SIP, 4620SIPCC, 4622, 4630, 9620, 9630 and 9650), the system displays the incoming caller information for the duration of the call. The system clears the caller information on the display for the user who puts the call on hold but retains the information on the display for the other users on the bridge. For outgoing calls, if the user on the bridge puts the call on hold, the display for the other user on the bridge is populated with the caller information. The default is n.
Maximum Number of Digits for Directed Group Call Pickup
Valid Entry 1 to 4
Usage
The maximum number of digits accepted for the pickup group number. The pickup group number is complete when it is followed by a # symbol. Default is 4.
PIN Checking for Private Calls
Enables or disables the PIN Checking for Private Calls feature that restricts users from making internal or external private calls by forcing them to enter a PIN code after dialing a PIN Feature Access Code (FAC).
Related links PIN Checking for Private Calls Using AAR Access Code on page 269 PIN Checking for Private Calls Using ARS Access Code on page 269
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Commercial Cost Avoidance (CCA) Related System Parameters
PIN Checking for Private Calls Access Code on page 268
Temporary Bridged Appearance on Call Pickup
Valid entries y
n
Usage
Enter y to allow a temporary bridged appearance for calls answered with the Call Pickup or Directed Call Pickup features. This field controls this capability on a system-wide basis.
Enter n to prevent the temporary bridged appearance of calls answered with these features.
Commercial Cost Avoidance (CCA) Related System Parameters
Backup MFSS CCA-ID
Use this field to search the PSTN or DSN mapping entries in the LDAP database. Enter a string without any white space.
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Feature-related system parameters
CCA Enabled
Use this field to enable Cost Commercial Avoidance (CCA).
Valid entry y n
Usage The system displays the CCA-related fields. The system does not display the CCA-related fields. The default value is n.
CTI Link Number
The CTI link number establishes a connection between Communication Manager and AES applications. Enter a CTI Link number of type ASAI-IP or ADJ-IP that is not used in any vector.
LSC CCA-ID
Use this field to search the PSTN or DSN mapping entries in the LDAP database. The default value of this field is the value populated in the Call Control Agent Identification (CCA-ID) field of MLPP. Though the value in the LSC CCA-ID field is from the Call Control Agent Identification (CCA-ID) field, you can change the value of the LSC CCA-ID field.
Note:
You can confirm the value of Call Control Agent Identification (CCA-ID) field on page 2 of the change system-parameters mlpp screen.
Primary MFSS CCA-ID
Use this field to search the PSTN or DSN mapping entries in the LDAP database. Enter a string without any white space.
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Chapter 55: Firmware Station Download
This screen downloads firmware to multiple stations of the same telephone type, either 2420 or 2410 DCP telephones. Downloads firmware to as many as 1000 stations per download schedule. You can schedule a specific time for the download, or you can administer the download to run immediately. Example command: change firmware station-download
Beginning Station
The first extension number in the range of telephones used to download the firmware. Up to 1000 stations can be included in a scheduled download. Accepts up to 16 digits.
Continue Daily Until Completed
Enables or disables the execution of a firmware download each day at the scheduled time until all specified telephones have received the firmware.
Download Set Type
Valid Entry 2410 DCP 2420 DCP
Usage The set type of DCP telephones to which firmware is to be downloaded.
Download Station Type
Type of station to which the firmware is to be downloaded.
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Firmware Station Download
Ending Station
The last extension number in the range of telephones used to download firmware. Up to 1000 stations can be included in a scheduled download. Accepts up to eight digits.
Schedule Download
Enables or disables a request to schedule a time for a firmware download to multiple DCP stations.
Source File
The name of the file used to retrieve the firmware download. Accepts up to 32 alphanumeric characters.
Related links File to Retrieve on page 884
Start Date/Time
Available only if Schedule Download is enabled.
Valid Entry
mm, dd, yyyy; hh, mm
Usage The month, day, year, and time for the firmware download to begin.
Related links Schedule Download on page 368
Stop Date/Time
Available only if Schedule Download is enabled.
Valid Entry
mm, dd, yyyy; hh, mm
Usage The month, day, year, and time for the firmware download to end.
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Related links Schedule Download on page 368
Stop Date/Time
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Chapter 56: Group Paging Using Speakerphone
Assigns digital speakerphones to a paging group. Users can page all the telephones in the group simultaneously by dialing the group extension. Example command: change group-page n, where n is the assigned group number.
Alert
Set this field to y for telephones that require an alert message to ring for an inbound call from a group page number. For example, Spectralink wireless telephones. Communication Manager does not activate the speakerphone of the Spectralink wireless telephones for a group page number. Therefore, the users do not hear the call ring from a group page number until the users manually answer the call.
Note:
Most of the Avaya telephones have the auto-answer capability, so if you set the Alert field to y, an Avaya telephone does not ring because the group paging initiates immediately.
Value y
n
Usage
Telephones, such as Spectralink wireless, receive an alert to ring for an inbound call from a group page number.
Telephones, such as Spectralink wireless, do not receive an alert to ring for an inbound call from a group page number. This is a default value.
ASAI
Value y
n
Usage
Enables ASAI events for paging groups. Enabling this functionality creates large spikes in the messaging traffic to the application.
ASAI events are not forwarded to the application for any call that has a paging group as parties on the call.
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COR
Valid Entry 0 to 995
COR
Usage A class of restriction (COR). To page the group, users' class of restriction must give them calling permission for the group's class of restriction.
Ext
Assigns a telephone extension to the group.
Group Extension
The extension users dial to page the members of this group.
Group Name
A name for the group that is informative to users. The callers' telephone displays when the caller pages the group. Accepts up to 27 alphanumeric characters.
Note:
Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Group Number
The identifying number the server running Communication Manager assigns to the group when it is created.
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Group Paging Using Speakerphone
Group Timeout (secs)
Valid Entry 0 to 600
Usage
Administer a timeout in seconds for the group page. After timeout, the paging party will be disconnected. The default value is 0 for no timeout.
Name
The name assigned to each extension in the group.
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
TN
The tenant number for this paging group. With tenant number, group paging can be partitioned by tenant.
Zip Tone Timer (secs)
If you set the value of the Zip Tone Timer (secs) field to 2 seconds, Communication Manager times out after this time and plays a Zip tone.
If you set the value in the Zip Tone Timer (secs) field to greater than 2 seconds and less than or equal to 10 seconds, Communication Manager tries to terminate every member of the group page within 2 seconds. If the attempt is successful, Communication Manager plays the Zip tone. Otherwise, Communication Manager sets the timer for the remaining time. After the second timer times out, Communication Manager plays the Zip tone and does not wait for any member of the group page to answer the call.
For example, if you set the value as 6 seconds, Communication Manager tries to terminate every member of the group page within 2 seconds. If the attempt is successful, Communication Manager plays the Zip tone. Otherwise, Communication Manager sets the timer for the remaining 4 seconds. After the 4 seconds timer times out, Communication Manager plays the Zip tone.
The range for this field is 2 through 10 seconds, and the default value is 7 seconds. If you set the value of this field between 2 through 6 seconds, the system displays the following warning
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Zip Tone Timer (secs)
message: Group page may not reach all members with timer value less than 7 sec.
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Chapter 57: Holiday Table
Defines individual holidays or holiday ranges. Example command: change holiday-table n, where n is the holiday table number.
Description
A phrase that describes the holiday. Accepts up to 27 characters.
End Day
Valid Entry 1 to 31
Usage The ending day of the holiday.
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End Hour
Valid Entry 0 to 23
Usage The ending hour of the holiday using a 24-hour clock.
End Min
Valid Entry 0 to 59
Usage The ending minute of the holiday.
End Month
Valid Entry 1 to 12
Usage The ending month of the holiday.
Name
The name of the holiday table. Accepts up to 27 characters.
Number
Valid Entry 1 to 10
Usage The holiday table number.
Start Day
Valid Entry 1 to 31
Usage The starting day of the holiday.
End Hour
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Holiday Table
Start Hour
Valid Entry 0 to 23
Usage The starting hour of the holiday using a 24-hour clock.
Start Min
Valid Entry 0 to 59
Usage The starting minute of the holiday.
Start Month
Valid Entry 1 to 12
Usage The starting month of the holiday.
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Chapter 58: Hospitality
Implements the system parameters associated with the hospitality features. Available only if Hospitality features are enabled for the system.
Example command: change system-parameters hospitality
Related links Hospitality (Basic) on page 834 Hospitality (G3V3 Enhancements) on page 835
Hospitality: page 1
Client Room Coverage Path Configuration
Indicates the server and the Property Management System (PMS) exchange coverage path information for guest stations.
Valid Entry act-nopms act-pms
Usage
The message is acknowledged (MESSAGE ACK), but no action is taken.
The server and PMS exchange and accept coverage path information. This field does not apply to normal PMS Protocol mode.
Controlled Restrictions Configuration
Indicates whether controlled restriction information is exchanged between the server and the PMS.
Valid Entry act-nopms act-pms
Usage The message is acknowledged (MESSAGE ACK), but no action is taken. The server and the PMS exchange and accept controlled restriction information.
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Hospitality
Default Coverage Path for Client Rooms
Applies only to stations with a client room class of service in the occupied mode. This field is used for transparent or ASCII mode. The value in this field is also used during a translation save as the coverage path for each station with client room class of service.
Valid Entry 1 to 9999 blank
Usage
The coverage path assigned when the server receives a check-out message for a valid extension or a new check-in.
Forward PMS Message to INTUITY Lodging
This field is used only in ASCII mode.
Valid Entry y n
Usage PMS-to-INTUITY messages are sent through the server. PMS-to-INTUITY messages are sent directly to the Avaya INTUITY Lodging system.
Housekeeper Information Configuration
Indicates whether housekeeper information is exchanged between the server and the PMS.
Valid Entry act-nopms act-pms
Usage The message is acknowledged (MESSAGE ACK), but no action is taken. The server and PMS exchange and accept housekeeper information.
Journal/Schedule Endpoint
Valid Entry Valid data extension number
PMS_LOG PMS_JOURNAL
Usage
A valid data extension number that is assigned to the data module connected to the Journal/Schedule printer. Cannot be a VDN extension. This extension can be the same as the PMS/ Log printer and both sets of reports can be printed on the same printer. This extension is dialed by the server to send journal information or schedule reports to the printer.
The printer is connected over a TCP/IP link, and this link is administered with a PMS_LOG service type for IP Services.
The printer is connected over a TCP/IP link, and this link is administered with a PMS_JOURNAL service type for IP Services.
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Related links Service Type on page 500
Hospitality: page 1
Message Waiting Configuration
Indicates whether message waiting notification requests and changes are being exchanged between the server and the PMS.
Valid Entry act-nopms act-pms
Usage
The message is acknowledged (MESSAGE ACK), but no action is taken.
Message waiting is active on the server and information between the PMS and server is transmitted.
Number of Housekeeper ID Digits
Valid Entry 0 to 6
Usage The number of digits that the housekeeper must dial for identification.
PMS Log Endpoint
Valid Entry Valid extension
PMS
Usage
The data extension number the server dials to access PMS. Cannot be a VDN extension.
The PMS is connected over a TCP/IP link. This link is administered with PMS service type for IP Services.
Related links Service Type on page 500
PMS LINK PARAMETERS
ASCII mode
Enables or disables ASCII-only mode used for the PMS message set. Available only with a transparent PMS protocol mode.
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Hospitality
PMS Link Maximum Retransmission Requests
Valid Entry 1 to 5
Usage
The number of times that the PMS can request acknowledgment for a message sent to the server.
PMS Link Maximum Retransmissions
Valid Entry 1 to 5
Usage
The number of times that the server retransmits a message to the PMS in response to a negative acknowledgment, or sends an inquiry for acknowledgment from the PMS before giving up on the message.
Milliseconds Before PMS Link Acknowledgment Timeout
Regulates how quickly the system responds to a message from the PMS, also known as "pace timing". This value is also used as the inquiry message (ENQ) time-out value. Should be kept as short as possible.
Valid Entry 100 to 1500
Usage
The time in milliseconds the system waits for an acknowledgment from the PMS indicating it correctly received a message.
PMS Endpoint
Valid Entry Valid extension
PMS
Usage
The data extension number the server dials to access PMS. Cannot be a VDN extension.
The PMS is connected over a TCP/IP link. This link is administered with PMS service type for IP Services.
Related links Service Type on page 501
PMS Protocol Mode
Valid Entry normal transparent
Usage
Indicates the message protocol mode used between the server and PMS. Coordinate this option with your PMS vendor.
Seconds Before PMS Link Idle Timeout
Valid Entry 5 to 20
Usage
The idle time in seconds that the server waits for an acknowledgment from the PMS before the server enters link failure mode from the PMS transmission link.
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Hospitality: page 2
Take Down Link for Lost Messages
Enables or disables taking down the PMS link if messages are getting lost. The PMS error log should be monitored if disabled.
Hospitality: page 2
Announcement Ports
Indicates the equipment location of two ports on the voice synthesizer circuit pack. Available only with a voice-synthesis announcement type.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
01 to 31 1 to 250 V1 to V9 01 to 31
Usage First and second characters are the cabinet number. Third character is the carrier. Fourth and fifth character are the slot number. Six and seventh characters are the circuit number.
Gateway Module Circuit
Related links Announcement Type on page 381
Announcement Type
Indicates the type of automatic wake up announcement the hotel guest receives.
Valid Entry external
integrated
Usage
Applicable when using an announcement adjunct. Requires entry of the circuit connection to the external announcement equipment.
Applicable when using the TN750B or TN750C announcement circuit pack. Requires entry of an extension for the integrated announcement.
Table continues...
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Hospitality
Valid Entry mult-integ
voice-synthesis music-on-hold silence
Usage
Multi-integrated is applicable when using the TN750B or TN750C announcement circuit pack. Automatic Wakeup can use integrated announcement circuit packs to play any one of multiple announcements to different extensions during a wake up call. Requires entry of an announcement extension.
A voice synthesis message is heard during the wake up announcement. Requires entry of port location information for the voice synthesizer circuit pack.
Uses the Music-on-Hold feature to provide the wake up announcement.
Silence is heard during the wake up announcement.
Related links Announcement Ports on page 381 Auxiliary Board for Announcement on page 382 Integrated Announcement Extension on page 384 Default Announcement Extension on page 383
Automatic Selection of DID Numbers
Enables or disables the Automatic Selection of DID Numbers for Guest Rooms feature. This feature assigns a two- to five-digit extension from a predetermined list of numbers to a hotel room telephone number that is disassociated with the room number.
Auxiliary Board for Announcement
The equipment location of an auxiliary trunk circuit that connects to the external announcement equipment. Available only for an external announcement type.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
01 to 31 1 to 250 V1 to V9 01 to 31
Usage First and second characters are the cabinet number. Third character is the carrier. Fourth and fifth character are the slot number. Six and seventh characters are the circuit number.
Gateway Module Circuit
Related links Announcement Type on page 381 Integrated Announcement Extension on page 384
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Default Announcement Extension on page 383
Hospitality: page 2
Custom Selection of VIP DID Numbers
The selection or removal of a DID number assigned to a room when a guest checks in. Available only if Automatic Selection of DID Numbers is enabled.
Related links Automatic Selection of DID Numbers on page 382
Daily Wakeup
Each extension permission can request daily wake up calls.
Default Announcement Extension
The default wake up announcement extension when using the integrated announcement circuit pack.
Available only with a multi-integrated announcement type.
Related links Announcement Type on page 381 Auxiliary Board for Announcement on page 382 Integrated Announcement Extension on page 384
Digit to Insert/Delete
Valid Entry 0 to 9
Usage
The current PMS message set uses the extension number as the room identifier. In many customer configurations, the leading digit of the extension number is dropped to screen the room number. To accommodate PMS devices that are based on room number and not extension. This leading digit can be deleted on messages from Avaya Communication Manager to the PMS, and then inserted back on messages from the PMS to Communication Manager.
Note:
The PMS interface supports three-, four-, or five-digit extensions, but prefixed extensions do not send the entire number across the interface. Only the assigned extension number is sent. Therefore, do not use prefixed extensions for numbers that are also going to use the Digit to Insert/Delete function.
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Hospitality
Display Room Information in Call Display
Indicates the type of guest room information displayed on telephone displays.
Valid Entry y
n
Usage
Telephones display the name and room number. The extension number and room number are not always the same number.
Telephones display the name and extension number.
Dual Wakeup
Allows or disallows each extension permission to request two wake up calls within one 24-hour period.
Extension to Receive Failed Wakeup LWC Messages
An extension that indicates where unsuccessful wake up LWC messages are stored. This is usually administered to an unassigned extension or to the attendant (attd). This extension cannot be a VDN extension. In addition, a LWC lamp for that extension is usually assigned to the attendant console as an indication of failed wake up calls.
Integrated Announcement Extension
The wake up announcement extension when using the integrated announcement circuit pack.
Available only for integrated announcement types.
Related links Announcement Type on page 381 Auxiliary Board for Announcement on page 382 Default Announcement Extension on page 383
Length of Time to Remain Connected to Announcement
Valid Entry 0 to 300
Usage
The length of time in seconds that a hotel guest is connected to an announcement. Applies only after the guest has heard the announcement completely one time, but continues to listen.
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Hospitality: page 2
Number of Digits from PMS
The number of digits being sent from the PMS to the server to identify room numbers.
Valid Entry 1 to 4 1 to 5 blank
Usage For normal mode For transparent or ASCII mode For mixed numbering in the server.
Number of Digits in PMS Coverage Path
Valid Entry 3 or 4
Usage The number of digits in the coverage path.
PMS Sends Prefix
Enables or disables PMS Sends Prefix to Indicate if the PMS sends a prefix digit to the server as part of the room numbering plan.
Room Activated Wakeup with Tones
Enables or disables the activation of wake up calls with tones. Wake up calls can be activated with tones that prompt users for the time they want to wake up. This allows room activated wake up calls without the use of a speech synthesizer or a display telephone.
Routing Extension on Unavailable Voice Synthesis
Valid Entry
Usage
Assigned extension A call is placed to this extension or to the attendant if a voice synthesis port is unavailable during voice synthesis entry of wakeup requests. This extension cannot be a VDN extension.
attd
An attendant group code.
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Hospitality
Time of Scheduled Emergency Access Summary Report
Valid Entry hh:mm:am/pm
Usage
The time of day that the Emergency Access Summary Report gets printed on the Journal/ Schedule printer.
Caution: Set the report for a time other than when the system does its scheduled maintenance tests.
Time of Scheduled Wakeup Activity Report
Valid Entry hh:mm:am/pm
Usage
The time of day that the Wakeup Activity Report gets printed on the Journal/ Schedule Printer. This report summarizes the wake up activity for each extension that had wake up activity for the past 24 hours.
Caution: Set the report for a time other than when the system does its scheduled maintenance tests.
Time of Scheduled Wakeup Summary Report
Valid Entry hh:mm:am/pm
Usage
The time of day that the Wakeup Summary Report gets printed on the Journal/ Schedule printer. This report gives an hour-by-hour summary of the number of scheduled wake up calls and a list of extensions to which wake up calls were attempted but were left incomplete during the hour.
Caution: Set the report for a time other than when the system does its scheduled maintenance tests.
VIP Wakeup
Allows or disallows permission for each extension to request VIP wake up calls.
VIP Wakeups Per 5 Minutes
Available only if VIP Wakeup is enabled for each extension.
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Hospitality: page 3
Valid Entry 1 to 50
Usage The number of VIP Wakeup calls allowed in a 5-minute interval.
Related links VIP Wakeup on page 386
Hospitality: page 3
ROOM STATES
Definition for Rooms in State 1 through 6
A definition for each room status. These definitions are only for Attendant Room Status. Accepts up to 30 characters. Example State 1 could be "clean, ready to use" and state 2 could be "occupied, needs cleaning".
HOSPITALITY FEATURES
Cancel Do-Not-Disturb for Wakeup Calls
Activates and deactivates whether a wakeup call will cancel Do-not-disturb.
Suite Check-in
Allows or disallows attendant's permission to have the system automatically check-in several related extensions with one check-in command.
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Chapter 59: Hunt Group
With Hunt groups, users or agents can answer calls at a predefined group of telephones or devices. Use the Hunt Group screen to create a hunt group, identified by a hunt group number, and to assign hunt group member users by their extension numbers. This screen can also be used to implement associated features such as Automatic Call Distribution (ACD) and Hunt Group Queuing. When a call comes into a hunt group, the system checks for the busy or idle status of extension numbers in the hunt group when answering. A Uniform Call Distribution (UCD) type hunt group selects the "most idle" extension in the group when answering a new call. A Direct Department Calling (DDC) type hunt group selects the first available extension (in the administered sequence) when answering a new call. Expert Agent Distribution (EAD), used only with Expert Agent Selection (EAS), selects the "most idle" agent or the "least occupied" agent with the highest skill level for the call's skill. From Release 8.1.3 onwards, if you enable (SA9142) - Redirected Display for SIP Hunt Group Members" field on the system-parameters special-applications screen, SIP station displays the caller's name and the hunt group name when a call is received on that SIP station. For more information, see Avaya Aura® Communication Manager Special Application Features document.
Note: Vector controlled splits/skills can be called directly through the split/skill extension instead of calling a VDN mapped to a vector that terminates the call to a vector controlled split/skill. However, the calls do not receive any announcements, be forwarded, redirect to coverage, or intraflow/interflow to another hunt group. Example command: change hunt-group n, where n is the assigned hunt group number.
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Hunt group: page 1
Hunt group: page 1
ACD
Indicates whether or not to use Automatic Call Distribution (ACD) for this hunt group. Available only if ACD is enabled for the system.
Valid Entry y
n
Usage
The hunt group functions as an ACD split/skill. Hunt groups used for voice messaging can function as ACD splits/skills.
The hunt group does not function as an ACD split/skill. This option should be used if this hunt group is on a remote server running Communication Manager and using voice messaging in a Distributed Communications System (DCS).
Related links ACD on page 842
(Calls Warning) Extension
Extension used by the Terminal Translation Initialization (TTI) feature to assign a port to this extension from the port itself. Once Calls Warning Port is assigned a valid port, then the extension is removed and unassigned.
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Hunt Group
This field cannot be blank. Available only if a queue has been enabled for this hunt group, and if a port number is not administered for the calls warning and time warning ports. Related links (Calls Warning) Port on page 390 Queue on page 396 (Time Warning) Port on page 397
(Calls Warning) Port
The seven-character port number assigned to connect the optional external Auxiliary Queue Call Warning Threshold lamp that flashes when the number of calls in queue has exceeded the queue warning threshold (assigned in Calls Warning Threshold). Available only if a queue has been enabled for this hunt group.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
01 to 31
Usage First and second characters are the cabinet number Third character is the carrier Fourth and fifth character are the slot number Six and seventh characters are the circuit number
This port is assigned to an Analog Line circuit pack or given an x designation if an extension is used.
Example
01A0612 is in cabinet 01, carrier A, slot 06, and circuit number (port) 12.
Related links Calls Warning Threshold on page 390 Queue on page 396
Calls Warning Threshold
Available only if a queue has been enabled for this hunt group.
Valid Entry 1 to 999
Usage
The number of calls that can be queued before the system flashes the queue status (feature buttons assigned on agents telephones) and the optional Auxiliary Queue Call Warning Threshold lamp assigned to the split/skill. These lamps are lighted steadily when at least one call is in queue and within the threshold. This value must be less than or equal to the queue length or left blank.
If you administer the Calls Warning Port field, do not leave this field blank.
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Related links Queue on page 396 (Calls Warning) Port on page 390
Hunt group: page 1
COR
Valid Entry 0 to 995
Usage
The class of restriction (COR) number that reflects the required restriction for the hunt group. If this is a hunt group supporting voice messaging in a Distributed Communications System (DCS), the CORs on this screen for each server running Communication Manager must be the same.
Coverage Path
The coverage path for the hunt group. Available only if the hunt group is not vector controlled.
Valid Entry digits
t1 to t999
Usage
A coverage path number. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Time of day table.
Related links Vector on page 398
Group Extension
An unused extension number assigned to the hunt group. This field cannot be blank.
Group Name
A character string that uniquely identifies the hunt group. Accepts up to 27 characters.
Note:
This field is supported by Unicode language display for the 4610SW, 4620SW, 4621SW, and 4622SW telephones. For more information on Unicode language display, see Administering Avaya Aura® Communication Manager. Example "parts dept", "purchasing", or "sales dept"
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Hunt Group
Related links Display Character Set on page 821
Group Number
The hunt group number.
Group Type
The group types of the hunt group depend on the system configuration. For more information about the availability of the group types, see the following table:
Hunt
circ
ddc
ucd-mia ead-mia ucd-loa ead-loa pad
sim
Group
and Call
Center
options
ACD = n, x
x
Queue =
n, Vector
= n
ACD = y, Vector = n or y, Skill = n
x
x
ACD = y, Vector = y, Skill = y
x
x
ACD = y, Vector = y, Skill = y, LOA = y
x
x
x
x
ACD = y, Vector = y, Skill = y, Business Advocat e = y
x
x
x
x
Table continues...
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Hunt group: page 1
Hunt
circ
ddc
ucd-mia ead-mia ucd-loa ead-loa pad
sim
Group
and Call
Center
options
ACD = y, Vector = y, Skill = y, Dynamic Advocat e = y
x
x
x
x
x
ACD = y,
x
Vector =
y, Skill =
y,
Service
Level
Maximize
r = y
For a configuration that is not listed, the LOA field and the Business Advocate or the Dynamic Advocate customer options are disabled and therefore, the configuration does not apply.
Valid entry circ
ddc
Usage
With the Circular call distribution, Communication Manager routes the calls to the administered extensions in a round-robin order. Communication Manager keeps track of the last extension in the hunt group to which a call was routed and routes the next call to the next extension in the circular list, independent of how long the extension has been idle.
You cannot use this option with ACD, Queues, or Vectors enabled.
The Circular option is available only with ACD disabled.
With the Direct Department Call (DDC) distribution, Communication Manager routes the calls to the first extension or ACD agent assigned in the ACD split. This option is also known as hot seat distribution. Communication Manager always routes an incoming call to the first available agent extension in the administered sequence. If the agent is unavailable, the call routes to the next available agent. Incoming calls are always routed to the first agent in the sequence, so calls are not evenly distributed among agents.
You cannot use this option when the hunt group is administered as a skill.
The DDC option is available only with the following configurations:
· ACD disabled, or
· ACD enabled, Split enabled or disabled, Vector enabled or disabled
Table continues...
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Hunt Group
Valid entry ucd-mia
ucd-loa
ead-mia
ead-loa
pad slm
Usage
With the Uniform Call Distribution-Most Idle Agent (UCD-MIA) call distribution, Communication Manager routes calls to the most-idle agent, based on the agent occupancy, and adds the agent who finishes the most recent call to the end of the idle list. You can use this option if the hunt group is used for voice messaging. Enter the ucd-mia or the ucd-loa group types when the system supports the Outbound Call Management feature and Controlling Adjunct type is asai-ip or adj-ip.
The UCD-MIA option is available only with ACD enabled.
With the Uniform Call Distribution-Least Occupied Agent (UCD-LOA) call distribution, Communication Manager routes calls to the least occupied agent, based on agent occupancy. You can use this option if the hunt group is used for voice messaging. Enter the ucd-mia or the ucd-loa group types when the system supports the Outbound Call Management feature and when the Controlling Adjunct type is asai-ip or adj-ip.
The UCD-LOA option is available only with LOA or Business Advocate enabled.
With the Expert Agent Distribution-Most Idle Agent (EAD-MIA) call distribution, Communication Manager routes calls to the agent with the highest skill level available for the call. If two or more agents with equal skill levels are available, Communication Manager routes the call to the most idle agent, based on when the agent finished the most recent call. The call is distributed to the agent best able to handle the call.
The EAD-MIA option is available only with EAS and Skill enabled.
With the Expert Agent Distribution-Least Occupied Agent (EAD-LOA) call distribution, Communication Manager routes calls to the agent available with the highest skill level for the call. If two or more agents are available with equal skill levels, Communication Manager routes the call to the least occupied agent, based on agent occupancy. The call is distributed to an agent best able to handle the call.
The EAD-LOA option is available only with EAS, LOA, or Business Advocate enabled.
With the Percentage Allocation Distribution (PAD) call distribution, Communication Manager selects an agent from a group of available agents, based on a comparison of the work time, and the target allocation for the skill of an agent.
The PAD option is available only with Dynamic Advocate enabled.
With the Service Level Maximizer (SLM) call distribution, Communication Manager compares the current service level for each SLM-administered skill to a user-defined call service level target and identifies the skills that are most in need of agent resources to meet their target service level. Communication Manager then identifies the available agents, assesses the overall opportunity cost of the agents, and selects only those agents whose other skills have the least need for their service at present.
The SLM option is available only with Service Level Maximizer enabled and Business Advocate disabled.
Related links ACD on page 842
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Business Advocate on page 842 Controlling Adjunct on page 400 Expert Agent Selection (EAS) on page 843
Hunt group: page 1
ISDN/SIP Caller Disp
This field is required for ISDN-PRI, ISDN-BRI, and SIP Trunks.
Valid Entry grp-name mbr-name blank
Usage The hunt group name is sent to the originating user. The member name is sent to the originating user. Displays the VDN name to the originating user.
Note:
Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Related links ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
Local Agent Preference
Enables or disables Local Agent Preference that routes an incoming ACD call to an idle agent by matching the location number of the incoming caller's station or trunk to the location number of an idle agent. Available only with Call Center Release 3.0 or later when Expert Agent Selection (EAS) and multiple locations are enabled for the system. Also, the hunt group must be administered as a skill hunt group.
Related links Call Center Release on page 843 Expert Agent Selection (EAS) on page 843 Multiple Locations on page 838 Skill on page 406
MM Early Answer
Enables or disables MM Early Answer. The system begins to answer an H.320 call and establish an audio channel before offering the conversion call to the hunt group. This starts billing for the call when the call is first put into queue. This field applies only for systems using Multimedia Call Handling.
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Hunt Group
Related links Multimedia Call Handling (Basic) on page 837
Night Service Destination
Unavailable for vector-controlled hunt group.
Valid Entry An assigned extension number
attd
Usage The destination where calls to this split redirect when the split is in the night service mode. This extension can be a VDN extension.
Must be a local extension for all features to work correctly. An attendant group code.
Related links Vector on page 398
Queue
Enables or disables a queue for the hunt group.
Queue Limit
Available only if a queue is enabled for the hunt group.
Valid Entry 1 to 999 unlimited
Usage
The limit to the number of calls that will queue.
The system dynamically allocates the queue slots from a common pool on an asneeded basis.
Related links Queue on page 396
Security Code
Use this field to enter the four-digit station security code for the Demand Print messages feature.
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Hunt group: page 1
SIP URI
Use this field to place a call using the URI. You can enter up to 70 characters. You can use letters, numbers, and special characters.
Following are not supported:
· Unicode characters · Eurofont characters · Using same URIs across hunt groups, VDNs, and stations within the same Communication
Manager. Example support-helpdesk@example.com
Time Warning Extension
Extension used by the Terminal Translation Initialization (TTI) feature to assign a port to this extension from the port itself. Once Time Warning Port is assigned a valid port, then the extension is removed and considered unassigned.
Available only if a queue is enabled for the hunt group. Required if a port number is not administered for the time warning port.
Related links Queue on page 396 (Time Warning) Port on page 397
(Time Warning) Port
The port number assigned to the Auxiliary Queue Time Warning lamp that flashes when the Time Warning Threshold has been reached by a call in queue. Available only if a queue has been enabled for the hunt group.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
01 to 31
Usage First and second characters are the cabinet number Third character is the carrier Fourth and fifth character are the slot number Six and seventh characters are the circuit number
This port is assigned to an Analog Line circuit pack or given an X designation if an extension is used.
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Hunt Group
Example 01A0612 is in cabinet 01, carrier A, slot 06, and circuit number (port) 12. Related links Queue on page 396 Time Warning Threshold on page 398
Time Warning Threshold
Available only if a queue is enabled for the hunt group and if a port number is not administered for the call warning and time warning ports.
Valid Entry 0 to 999
Usage
The time in seconds that a call can remain in the queue before the system flashes the Queue status lamps (feature buttons assigned members telephones) and the Auxiliary Queue Time Warning lamp assigned to this split/skill. An entry of 0 provides a warning whenever a call is queued.
Related links Queue on page 396 (Time Warning) Port on page 397
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
Vector
Enables or disables this hunt group as vector controlled. Available only if Basic Vectoring is enabled for the system.
Related links Vectoring (Basic) on page 845
Hunt group: page 2
This screen can vary according to values for particular fields on the previous page. If the ACD is disabled for the system, this page is omitted.
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Hunt group: page 2
AAS
Enables or disables this hunt group serving as an Auto-Available Split (AAS). With AAS, members of an ACD split or skill can be in auto-in work mode continuously. An agent in auto-in work mode becomes available for another ACD call immediately after disconnecting from an ACD call. Available only if ACD is enabled for this hunt group.
Related links ACD on page 389
Adjunct CTI Link
Use this field to assign a CTI link to the hunt group. You must not leave this field blank. The system displays this field only if ACD is enabled for the hunt group and the Controlling Adjunct field shows asai-ip or adj-ip.
Related links ACD on page 389 Controlling Adjunct on page 400
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Hunt Group
Controlling Adjunct
Use this field to specify the controlling adjunct. The system displays this field only if ACD is enabled for the hunt group.
Valid entry
none
asai-ip or adj-ip
Usage
Members of the hunt group are not controlled by an adjunct processor. Members of the hunt group are controlled by an ASAI-IP or ADJ-IP link.
Related links ACD on page 389 ASAI Link Core Capabilities on page 829 Computer Telephony Adjunct Links on page 831
Dynamic Percentage Adjustment
Enables or disables automatic adjustments to agents' target allocations as needed to help meet the administered service level targets. Available only if ACD is enabled for the hunt group and this is a Percent Allocation Distribution (PAD) hunt group. Requires Business Advocate software.
Related links ACD on page 389 Business Advocate on page 842 Group Type on page 392
Dynamic Queue Position
Enables or disables dynamic queue operation to the calls queued to the skill. Dynamic Queue Position is a Business Advocate feature for calls to queue from multiple VDNs to a single skill, while maintaining different service objectives for those VDNs. Available only if ACD, Expert Agent Selection (EAS), and Skill are enabled for the hunt group. Requires Business Advocate software.
Related links ACD on page 389 Business Advocate on page 842 Skill on page 406
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Dynamic Threshold Adjustment
Enables or disables automatic adjustments to overload thresholds to engage reserve agents a bit sooner or a bit later to meet the administered service levels. Available only if ACD and Service Level Supervisor are enabled for the hunt group. Requires Business Advocate software.
Related links ACD on page 389 Business Advocate on page 842 Service Level Supervisor on page 404
Expected Call Handling Time (sec)
Available only if ACD is enabled for the system and if either Vectoring (Advanced Routing) or Business Advocate is enabled for the system.
Valid Entry
1 to 9999 in increments of 1
Usage
Establishes the number of seconds for expected call handling. This value is used to initialize Expected Wait Time and is also used by the Business Advocate Percent Allocation feature.
Related links ACD on page 389 Business Advocate on page 842 Vectoring (G3V4 Advanced Routing) on page 846
Inflow Threshold (sec)
Available only if ACD and a queue are enabled for the system. unavailable for a vector-controlled hunt group.
Valid Entry 0 to 999
Usage
The number of seconds that a call can remain in the queue before no more calls are accepted by the queue. If 0 is entered, a call is redirected to this split/skill only if there is an available agent.
Related links ACD on page 389 Queue on page 396 Vector on page 398
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Hunt Group
Level 1 Threshold (sec)
Available only if ACD and Service Level Supervisor are enabled for the hunt group.
Valid Entry 0 to 60
Usage The number of seconds for the first Expected Wait Time (EWT) threshold.
Example If there are 45 calls whose EWT exceeds 45 seconds, threshold 1 will have been exceeded.
Related links Service Level Supervisor on page 404
Level 2 Threshold (sec)
Available only if ACD and Service Level Supervisor are enabled for the hunt group.
Valid Entry 0 to 60
Usage The number of seconds for the second Expected Wait Time (EWT) threshold.
Example
If there are 60 calls whose EWT exceeds 60 seconds, threshold 2 will have been exceeded.
Related links ACD on page 389 Service Level Supervisor on page 404
Maximum Auto Reserve Agents
Available only if ACD is enabled for the hunt group and it is a Service Level Maximizer (SLM) type hunt group.
Valid Entry 0 to 9
Usage
The maximum number of Auto Reserve Agents available for this skill (hunt group). Any time an auto-reserve skill is in danger of falling below its target service level percent, some of this skill's agents are auto-reserved (kept idle in other skills) so that they are available when a new call arrives for this skill. Default is 0.
Related links Group Type on page 392
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Hunt group: page 2
Measured
Provides measurement data for the ACD split/skill to VuStats or BCMS. Available only if ACD is enabled for the hunt group and VuStats or BCMS is enabled for the system.
Valid Entry internal
external
both none
Usage
Provides measurements made by the Call Management System that are internal to the server running Communication Manager.
Provides measurements made by the Call Management System that are external to the server running Communication Manager.
Provides measurements collected both internally and externally.
Measurement reports for this hunt group are not required.
Related links ACD on page 389 BCMS (Basic) on page 842 VuStats on page 847
Multiple Call Handling
Defines whether the hunt group can have multiple call handling capabilities, and if so, what type. Available only if ACD is enabled for the hunt group and Multiple Call Handling is enabled for the system.
Valid Entry none on-request
many-forced
one-forced
one-per-skill
Usage
Agents who are members of that split/skill can only receive an ACD call from that split/skill when the telephone is idle.
Agents in the Multiple Call Handling split/skill can place a non-ACD or an ACD call on hold and select an available work mode. A queued ACD split/ skill or direct agent call then is routed to the agent.
An ACD call is delivered automatically to an idle line appearance if the agent is in the Auto-In/Manual-In (MI/AI) work mode and an unrestricted line appearance is available.
An ACD call is delivered automatically to an idle line appearance if the agent has no other ACD call on the station, is in the Auto-In/Manual-In (MI/AI) work mode, and an unrestricted line appearance is available.
An ACD call is delivered automatically to an idle line appearance if the agent has no other ACD call for that skill on the station, is in the Auto-In/ Manual-In (MI/AI) work mode, and an unrestricted line appearance is available. Valid in an EAS environment for a skill-enabled hunt group.
Related links ACD on page 389
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Hunt Group
Multiple Call Handling (Forced) on page 844 Skill on page 406
Priority On Intraflow
Enables or disables having calls intraflowing from this split to a covering split and given priority over other calls waiting in the covering split queue. Available only if ACD field is enabled for the hunt group. Unavailable for a vector-controlled hunt group.
Related links ACD on page 389 Vector on page 398
Service Level Interval
The time interval when Actual Service Level (ASL) calculations run. ASL is one of the Service Level Maximizer (SLM) algorithms used for most situations, particularly for low staff or low traffic. The interval can be set to the same interval used when specifying the target objectives for the application. Available only if Actual is administered for the SLM algorithm feature and this is an SLM-type hunt group.
Valid Entry hourly
daily
weekly
Usage
ASL algorithm calculations for accepted call and total call components are set to 0 at hourly intervals.
ASL algorithm calculations for accepted call and total call components are set to 0 at daily intervals. This is the default.
ASL algorithm calculations for accepted call and total call components are set to 0 at weekly intervals. The weekly interval starts as 00:00 hours on Sunday.
Related links Service Level Maximizer Algorithm on page 337 Group Type on page 392
Service Level Supervisor
Enables or disables Service Level Supervisor for this skill. Service Level Supervisor is a Business Advocate feature that alleviates the need to move agents from skill to skill during emergencies or unanticipated peaks in call volume. Available only if ACD is enabled for the hunt group and this is an Expert Agent Selection (EAS) skill hunt group. Requires Business Advocate software.
Related links ACD on page 389
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Skill on page 406 Business Advocate on page 842
Hunt group: page 2
Service Level Target (% in sec)
The system displays Service Level Target (% in sec) when the ACD field and the Measured field is not blank, and when one or more of the following features are active:
· BCMS/VuStats Service Level field on the System Parameters Customer-Options screen is active and the Measured field is set to internal or both. The service level target in seconds is used as the acceptable level for reporting the percentage of calls answered within the specified time. The percentage can be set to the default of 80%.
· Business Advocate on the System Parameters Customer-Options screen is active. The service level target in seconds is used for the Business Advocate Service Level Supervisor service level objective. This service level target can also be used for the dynamic percentage adjustment when the Dynamic Threshold Adjustment field on the Hunt Group screen is y and for the dynamic percent adjustment when the Group Type field on the Hunt Group screen is pad and the Dynamic Percent Adjustment field on the Hunt Group screen is set to y.
· The system displays the Service Level Target field when the Group Type field on the Hunt Group screen is slm, and on the System Parameters Customer-Options screen, the Service Level Maximizer is active, and the Business Advocate customer option license is inactive. In this case the setting is also used as the service level target to trigger SLM.
· Interruptible Aux Threshold field on the Hunt Group screen is set to service-level-target. The Interrupt Aux feature is triggered if the service level drops below the administered percentage of calls in the specified seconds.
Valid Entry
1 to 99 (percentage)
1 to 9999 (time in seconds)
Usage The percentage component of the service level target. The default value is 80%.
The time component of the service level target. The default value is 20 seconds.
Related links ACD on page 389 BCMS (Basic) on page 842 Business Advocate on page 842 Dynamic Percentage Adjustment on page 400 Dynamic Threshold Adjustment on page 401 Group Type on page 392 Interruptible Aux Threshold on page 409 Service Level Maximizer on page 844 VuStats on page 847
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Hunt Group
Service Objective
Available only if ACD is enabled for the hunt group and this is an Expert Agent Selection (EAS) skill hunt group. Requires Business Advocate software.
Valid Entry 1 to 9999
Usage
Sets a service objective for a specific skill as the number of seconds the call gets answered. The default value is 20.
Related links ACD on page 389 Business Advocate on page 842 Skill on page 406
Skill
Enables or disables this hunt group as an Expert Agent Selection (EAS) skill. Available only if ACD is enabled for the hunt group and EAS is enabled for the system.
Related links ACD on page 389 Group Type on page 392 Expert Agent Selection (EAS) on page 843
SLM Count Abandoned Calls
Available only if Actual is the administered Service Level Maximizer (SLM) algorithm for the feature and this is an SLM-type hunt group.
Valid Entry y
n
Usage
Abandoned calls are included in the Actual Service Level (ASL) algorithm calculations for SLM.
Abandoned calls are excluded in the ASL algorithm calculations for SLM. This option is best used when reporting for this application does not account for calls that are abandoned while in skill queues.
Related links Group Type on page 392 Service Level Algorithm for SLM on page 343
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Hunt group: page 2
Supervisor Extension
The extension number of the ACD split/ skill supervisor that agents reach when using the Supervisor Assist feature. The extension number cannot be a Vector Directory Number (VDN).
Available only if ACD field is enabled for the system.
Related links ACD on page 389
Timed ACW Interval (sec)
Available only if ACD is enabled for the hunt group and Timed ACW is enabled for the system.
Note: This field can be overridden by the settings administered for a vector. Coordinate the settings when setting up delays.
Valid Entry 1 to 9999
Usage
The number of seconds an agent in auto-in work mode remains in After Call Work (ACW) mode after a call drops. After this time interval expires, the agent automatically becomes available. Timed ACW cannot be administered if the hunt group is adjunct controlled, is an AUDIX Message Center, or is an auto-available split.
Related links ACD on page 389 Timed ACW on page 845
VuStats Objective
Available only if ACD is enabled for the hunt group and VuStats is enabled for the system. Also, the hunt group must be administered to collect internal or both internal and external measurement data for VuStats.
Valid Entry 0 to 99999
Usage
A numerical objective. An objective is a split or skill goal for the call. This could be an agent objective such as a specific number of calls handled or an average talk time. The objective could also be a percent within the service level. The VuStats screen displays the objective. The agents and supervisors can compare the current performance against the value of the objective for the split or skill.
This value applies to customized VuStats display formats.
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Hunt Group
Related links ACD on page 389 Measured on page 403 VuStats on page 847
Hunt group: page 3
This screen can vary according to values for particular fields on the previous page. If the ACD is disabled for the system, this page is omitted.
Forced Entry of Stroke Counts or Call Work Codes
Enables or disables the requirement that either a Stroke Count or Call Work Code must be entered for each call answered by an agent when in the Manual-In mode. Available only if ACD is enabled for the hunt group and if the hunt group does not have a Controlling Adjunct.
Related links ACD on page 389 Controlling Adjunct on page 400
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Hunt group: page 3
Interruptible Aux Deactivation Threshold
This field specifies the maximum level of service above which the system deactivates the Interruptible Aux feature.
Use the Interruptible Aux threshold and the Interruptible Aux Deactivation Threshold fields to maintain a buffer between the two levels of service that trigger the Interruptible Aux feature. The values of the Interruptible Aux threshold and the Interruptible Aux Deactivation Threshold fields must have a difference of one unit.
The valid entries for this field depend on the value of the Interruptible Aux threshold field:
· If you set the value of the Interruptible Aux threshold field to calls-warning-threshold, the valid entries for the Interruptible Aux Deactivation Threshold field are from 0 to 998.
· If you set the value of the Interruptible Aux threshold field to service-level-target, the valid entries for the Interruptible Aux Deactivation Threshold field are from 0 to 100.
· If you set the value of the Interruptible Aux threshold field to time-warning-threshold, the valid entries for the Interruptible Aux Deactivation Threshold field are from 0 to 998.
· If you set the value of the Interruptible Aux threshold field to none, the system does not display the Interruptible Aux Deactivation Threshold field.
Interruptible Aux Threshold
This field specifies the minimum level of service below which the system enables the Interruptible Aux feature. Use the Interruptible Aux feature to make the Expert Agent Selection (EAS) agents in the AUX work mode available to receive calls. This feature works only if the agents have an interruptible reason code.
Valid entry service-level-target
calls-warningthreshold
time-warningthreshold
none
Usage
Use this option to enable the Interruptible Aux feature when the service level drops below the administered percentage of calls within the specified period. For example, if you set the Service Level Target (% in sec) field to 90% of calls in 30 seconds, the system enables the Interruptible Aux feature if the service drops to 89% of calls in 30 seconds.
Use this option to enable the Interruptible Aux feature when the number of calls in the queue for a hunt group exceeds the specified number of calls. For example, if you set the Calls Warning Threshold field to 20, the system enables the Interruptible Aux feature if the number of calls in the queue exceeds 20.
Use this option to enable the Interruptible Aux feature when the oldest call is in the queue for longer than the specified number of seconds. For example, if you set the Time Warning Threshold field to 60 seconds, the system enables the Interruptible Aux feature if the oldest call is in the queue for more than 60 seconds.
Use this option if you do not want to administer the Interruptible Aux feature.
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Hunt Group
Redirect on IP/OPTIM Failure to VDN
A blank in the field redirects the call back to the hunt group. VDN extension redirects to the specified VDN.
If Redirect on IP/OPTIM Failure to VDN is not assigned, the call is re-queued to the same skill at a high priority. If there are no queue slots available, the caller will hear a busy signal. If all fails, the caller receives ring back until the system receives a caller disconnect.
Related links ACD on page 389
Redirect on No Answer (rings)
Available only if ACD is enabled for the hunt group.
Valid Entry 1 to 20
blank
Usage
The maximum number of rings before a call redirects back to the split/skill, or to the administered VDN.
Deactivates Redirect on No Answer.
Related links ACD on page 389
Redirect on No Answer to VDN
The extension number of the VDN used to redirect a Redirect On No Answer (RONA) call to a VDN instead of to the split/skill. The administered VDN must be on-premises and must be administered on the system. The VDN can specify a vector that routes to an off-premises VDN.
Direct Agent calls go to the agent's coverage path if it is administered. If not, the calls go to a VDN.
Available only if ACD is enabled for the hunt group. Requires administration of the number of rings before a call will redirect.
Related links ACD on page 389 Redirect on No Answer (rings) on page 410
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Hunt Group: page 4
Hunt Group: page 4
The Hunt Group Message Center screen can vary according to system configuration and values populating particular fields.
AUDIX Name
The name of the AUDIX machine. Must be the same name as the IP Node name and administered after the IP Node is configured.
Related links IP Node Names on page 475 IP Node Names on page 475
Calling Party Number to INTUITY AUDIX
The system displays Calling Party Number to INTUITY AUDIX when the Message Center field is audix or rem-vm.
Valid entries y/n
Usage Enter y to send the calling party's number to INTUITY AUDIX®
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Hunt Group
Related links Message Center on page 413
First Announcement Delay (sec)
Available only if a queue is administered for the hunt group. Unavailable if the hunt group is vector controlled.
Valid Entry 0 to 99
blank
Usage
The number of seconds that a call remains in queue before the associated first announcement is given the calling party. The call retains its place in the queue while the caller is listening to the recorded announcement. If the call has not been answered after the announcement, the caller hears music for first announcement only if Music-on-Hold is provided, or ringing for as long as the call remains in queue.
When 0 is entered, the first announcement is provided immediately to the caller. This value is set automatically to 0 if there is no queue.
There is no first announcement.
Related links Queue on page 396 Vector on page 398
First Announcement Extension
The recorded announcement extension number the caller receives after being in the queue for the time interval specified in First Announcement Delay. If the call hasn't been answered after the announcement, the caller hears music only after the first announcement if Music-on-Hold is provided, or ringing for as long as it remains in the queue. If this is the forced first announcement, the caller always hears ringback after the announcement. Otherwise, the caller hears music, if provided.
Available only if ACD and a queue is administered for the hunt group. Unavailable if the hunt group is vector controlled.
Note:
If you enter a shortened extension of the multilocation dial plan, the system does not perform certain administration and validation tasks. Therefore, the system might not display the resultant warnings or submittal denials.
Related links ACD on page 389
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Hunt Group: page 4
LWC Reception
Use this field to specify the location where the system must store the LWC messages.
Valid entry spe none audix
Usage Use this option to store the LWC messages on Switch Processor Element (SPE). Use this option if you do not want to store the LWC messages. Use this option to store the LWC messages on the voice messaging system.
Related links AUDIX Name on page 411
Message Center
The type of messaging adjunct for the hunt group. For information on system capacities, see Avaya Aura® Communication Manager System Capacities Table.
Valid Entry audix fp-mwi
msa msa-vm rem-vm qsig-mwi sip-adjunct none
Usage AUDIX located on this server running Communication Manager Public network allowing AUDIX to be located on another switch. Available only if ISDN Feature Plus is administered for the system. Messaging Server Adjunct A voice mail system integrated using Mode Codes or Digital Station Emulation DCS feature allowing voice mail to be located on another server QSIG network allowing voice mail to be located on another server SIP message center server The hunt group does not serve as a message hunt group.
Related links ISDN Feature Plus on page 836
Message Center AUDIX Name
The name of the Message Center AUDIX. Available only if the messaging type is audix or rem-vm. Related links
Message Center on page 413
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Hunt Group
Message Center MSA Name
The name of the Message Center Messaging Server Adjunct (MSA). Available only if the messaging type is msa.
Related links Message Center on page 413 Message Center AUDIX Name on page 413
Primary
Available only if the Message Center field is audix or rem-audix.
Valid entries y/n
Usage Enter y to indicate that the specified messaging server is the primary adjunct.
Related links Message Center on page 413
Provide Ringback
Enables or disables ringback to the calling party until a Connect is received for the call to the Messaging system. Ringback is discontinued upon receipt of the Connect indication. Used for an SBS trunk for the QSIG MWI hunt group. A call covering to the message center provides ringback to the caller during the coverage interval. Available only if the messaging type is fp-mwi or qsigmwi.
Related links Message Center on page 413
Routing Digits (e.g. AAR/ARS Access Code)
A one- to four-digit AAR (qsig-mwi) or ARS (fp-mwi) access code. This access code is prepended to the AUDIX Complete Number to define a route to the Message Center switch hunt group containing the line ports to the AUDIX. Accepts characters * and #.
Available only if the messaging type is qsig-mwi or fp-mwi.
Related links Message Center on page 413
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Hunt Group: page 4
Second Announcement Delay (sec)
Available only if ACD and queues are enabled for the hunt group. Unavailable if the hunt group is vector controlled.
Valid Entry 1 to 99
blank
Usage
The time in seconds before the call in the queue receives a second recorded announcement or that the second announcement is repeated. If this split/skill or hunt group is a coverage point for another split/skill, this delay should not be more than 15 seconds.
There is no second announcement.
Related links ACD on page 389 Queue on page 396 Vector on page 398
Second Announcement Extension
The extension number assigned to a second recorded announcement. Left blank if there is no second announcement.
Available only if ACD and queues are enabled for the hunt group. Unavailable if the hunt group is vector controlled.
Note:
If you enter a shortened extension of the multilocation dial plan, the system does not perform certain administration and validation tasks. Therefore, the system might not display the resultant warnings or submittal denials.
Related links ACD on page 389 Queue on page 396 Vector on page 398
Second Announcement Recurring
Allows or disallows repeating the second announcement. Available only if ACD and queues are enabled for the hunt group. Unavailable if the hunt group is vector controlled.
Related links ACD on page 389 Queue on page 396
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Hunt Group Vector on page 398
Send Reroute Request
Allows or disallows rerouting getting invoked when a call covers through a qsig-mwi hunt group. Available only if the messaging type is qsig-mwi and Supplementary Services with Rerouting is enabled for the system.
Related links Message Center on page 413 Supplementary Services with Rerouting on page 849
TSC per MWI Interrogation
Controls Temporary Signaling Connections (TSCs) used for message waiting interrogations for users that are "local" to the system in which the hunt group is administered. Available only if the messaging type is qsig-mwi.
Valid Entry y
n
Usage
Communication Manager brings the TSC up, executes the Interrogate operation, and then tears the TSC down.
Communication Manager uses the existing TSC sending FACILITY messages to request MWI status if the TSC is already set up, or sets up a TSC. When the interrogation operation is complete, leaves the TSC up, subject to the existing timer. This is the default.
Related links Message Center on page 413
Voice Mail Extension
The UDP extension of the voice mail hunt group on the host server running Communication Manager.
Available only if the messaging type is rem-vm.
Related links Message Center on page 413
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Hunt Group: page 5 through X
Voice Mail Handle
The Session Manager (SM) handle that can receive voice mail. Can be left blank if a Voice Mail Number has been assigned.
Voice Mail Number
The 1- to 17-digit voice mail dial-up number. The qsig-mwi selection shows the complete number of the AUDIX hunt group on the Message Center server for QSIG MWI. The fp-mwi selection shows the public network number of the AUDIX hunt group on the Message Center server. Available only if Basic Call Setup and Basic Supplementary Services are enabled for the system, and the messaging type is qsig-mwi or fp-mwi. Related links Basic Call Setup on page 848 Basic Supplementary Services on page 848 Message Center on page 413
Hunt Group: page 5 through X
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Hunt Group
Administered Members (min/max)
The minimum and maximum member number administered for this hunt group. Available for all member pages.
At End of Member List
Displays the current page as also the last page.
Group Extension
The extension of the hunt group.
Group Number
Displays the number of the hunt group.
Group Type
Displays the type of hunt group.
Member Range Allowed
The range of allowed members. These values vary depending on the system or configuration.
More Members Exist
Statement that there are more members and more pages than currently displayed.
Total Administered Members
The total number of members administered for the hunt group.
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Hunt Group: page 5 through X
GROUP MEMBER ASSIGNMENTS
Ext
The assigned station or attendant console extension. This extension cannot be a Vector Directory Number (VDN). The data module cannot be a member of an ACD split/skill. Administers the assigned station or attendant console extension only if the controlling adjunct is administered as none. Displays the assigned station or attendant console extension if the controlling adjunct is administered as asai-ip or adj-ip.
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly. Related links Controlling Adjunct on page 400
Name
The name assigned to the extension number when it is administered in the system. Related links
Ext on page 419
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Chapter 60: Incoming Call Handling Treatment
Incoming Call Handling Treatment (ICHT) specifies a unique call treatment for incoming calls on ISDN and SIP trunk groups. This screen is available only if digit handling is administered as overlap, and the direction is set to outgoing.
Note: ICHT is also used to modify the Calling Party number when Communication Manager is used as a feature server. The Incoming Call Handling Treatment Table screen displays entries with a predefined service or feature before entries with a user-defined service or feature. The user-defined entries are always listed in the reverse order on Incoming Call Handling Treatment Table as compared to the entries that show on Network Facilities. Therefore, for two user-defined services or features, ABC and XYZ, you can force XYZ to show before ABC on Incoming Call Handling Treatment Table by administering XYZ after ABC on Network Facilities. Note: DCS features that use the remote-tgs button on the remote server do not work when the local trunk group deletes or inserts digits in the incoming call. To change the value of the fields on the Incoming Call Handling Treatment Table, use the following command: change inc-call-handling-trmt trunk-group n, where n is an ISDN or a SIP trunk group.
Called Len
Valid Entry 0 to 21
blank
Usage
The number of digits received for an incoming call. Zero is used when the Public Switched Telephone Network (PSTN) provider does not provide any "Number Digits" within the received Called Party IE, such as in Japan.
When Called Number has also been set to blank, so that any length of digits associated with the Called Party IE of the Incoming SETUP message matches this field.
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Called Number
Called Number
Valid Entry 1 to 16 blank
Usage Specifies the leading digits received for an incoming call. Matches any number associated with the specified service or feature.
Del
Valid Entry 1 to 21 blank
Usage
The number of leading digits to be deleted from the incoming Called Party Number. Calls of a particular type can be administered to be routed to a single destination by deleting all incoming digits and then administering the Insert field with the required extension.
Insert
Valid Entry 1 to 16 * #
Usage
The number of digits prepended to the front of the remaining digits after any optional digit deletions have been performed. The resultant number formed from digit deletion and insertion is used to route the call, provided night service is not in effect.
Per Call CPN/BN
Specifies when and how to request Calling Party Number (CPN) or Billing Number (BN) for calls of this type. Available only with ISDN trunk groups.
Valid Entry cpn-only bn-only bn-pref cpn-pref none
Usage Calling party number only Billing number only Prefers billing number, but accepts calling party number Prefers calling party number, but accepts billing number Communication Manager will not request either CPN or BN for any incoming calls of this type.
Table continues...
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Incoming Call Handling Treatment
Valid Entry blank
Usage
Leave blank when connected to another media server or switch, or when connected to a public network outside North America. Within North America, leave blank when connected to a public network that prohibits customer equipment to request CPN or BN for individual incoming calls. The AT&T Switched Network offers this service under the titles "CPN/BN to Terminating End on a Per-Call Basis" and "ANI (BN) on Request".
Note:
A 4-second delay occurs in terminating the call to the far-end station if the connecting server or switch stops responding to the request.
Night Serv
Available only for ISDN trunk groups.
Valid Entry
Usage
Assigned extension Specifies a night service extension per Service/Feature. A Vector Directory Number
Attendant group access code
(VDN) is acceptable. This entry is overridden by the night service administered for an individual trunk or the trunk group.
attd
blank
Related links Night Service on page 911 Night on page 546
Service/Feature
Displays the number that corresponds to the server type administered for the trunk group or network facility. Administration is required for a call-by-call service type.
Valid Entry
Usage
Valid pre-defined Service/Feature values for cbc trunk groups
These values are administered for Network Facilities.
Table continues...
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Valid Entry
User-defined facility type administered for the service type
Usage · 0 = feature · 1 = service · 2 = incoming · 3 = outgoing
Related links Network Facilities on page 630 Facility Type on page 630
Service/Feature
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Chapter 61: Integrated Announcement Boards
Moves integrated announcement boards that have been previously administered on the Announcements/Audio Sources screen to a new board location. Displays a list of all administered integrated announcement circuit packs.
Example command: display integrated-annc-boards
display integrated-annc-boards INTEGRATED ANNOUNCEMENT SOURCES
Page 1 of 13
Tim
Num Chk
Src Sfx Rem Rt Annc Sum
Tim
Num Chk
Src Sfx Rem Rt Annc Sum
1: M1
NA NA 1 NA
16:
2: 001V9 AP 1372 64 1 NA
17:
3:
18:
4:
19:
5:
20:
6:
21:
7:
22:
8:
23:
9:
24:
10:
25:
11:
26:
12:
27:
13:
28:
14:
29:
15:
30:
Board Location
The physical location of the integrated announcement circuit pack (UUCSS).
Chk Sum
This field displays the Checksum ID. This field applies to only TN750 and is not applicable to VAL.
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Last Board Location Saved
Last Board Location Saved
Displays last board location saved. Applies to TN750 only; not applicable to VAL.
Number of Recordings
The number of non-zero-length announcement recordings or files on the circuit pack.
Rate
The announcement's compression rate.
Sfx
The circuit pack suffix letters.
Time Remaining
The amount of recording time in seconds remaining on the circuit pack at the 64Kb rate.
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Chapter 62: Integrated Announcement Translations
Changes board locations currently administered on the Announcements/ Audio Sources screen to a new board location. Example command: change integ-annc-brd-loc
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Change all board location translations from board
Change all board location translations from board
Valid Entry board cabinet 1 to 3 carrier A to E slot 1 to 20 or gateway 1 to 10 module V1 to V9
Usage The VAL board that is currently administered.
to board
Use this field to specify the VAL board for announcement translations. You can use one of the following ways to enter a value in this field:
· board number · cabinet 1 to 3: carrier A to E: slot 1 to 20 · gateway 1 to 10: module V1 to V9
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Chapter 63: Intercom Group
Assigns extensions to intercom groups. Example command: change intercom-group n, where n is the assigned intercom group number.
DC
Assigns a dial code to an extension. Accepts up to two digits. The dial code is the code users must dial to make intercom calls to the corresponding extension. The number of digits entered must exactly match the number administered for the length of the dial code. This field cannot be blank. Example If the length of the dial code is set to 2, type 1 as 01 in the DC field. Related links Length of Dial Code on page 429
Ext
Assigns an extension to the group. A Vector Directory Number (VDN) cannot be used as an extension.
Group Number
The group ID number.
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Length of Dial Code
Length of Dial Code
Sets the number of digits that users must dial to access an extension in the group.
Valid Entry 1 2
Usage For nine or fewer members. For 10 or more members.
Name
The name associated with the extension that has been administered to join the group.
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly. Related links Name on page 750
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Chapter 64: Inter-Exchange Carrier (IXC) Codes
Identifies the IXC in the Call Detail Recording (CDR). Example command: change ixc-codes
Inter-Exchange Carrier (IXC) Codes: page 1
IXC Access Number
The digits dialed or inserted by AAR/ARS into the outpulsed digit string to access the interexchange carrier. You cannot use duplicate access numbers in the table. Accepts from 2 to 11 digits and the * character.
IXC Name
A description that identifies the IXC. Accepts up to 15 characters.
Inter-Exchange Carrier (IXC) Codes: page 2
IXC Code Format
A one- to four-digit IXC code format. Includes *, x, X, xxxx (for line 1), and xxx (for line 2).
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Inter-Exchange Carrier (IXC) Codes: page 2
IXC Prefix
A one- to three-digit prefix. Includes *, 101 (for line 1) and 10 (for line 2).
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Chapter 65: Intra-Switch CDR
Administers extensions for which Intra-Switch Call Detail Recording (CDR) is enabled. Note: Attendants are disallowed to be optioned for the Intra-Switch CDR feature.
Example command: change intra-switch-cdr n, where n is the assigned extension number.
Assigned Members
The number of extensions currently administered for Intra-switch CDR.
Extension
The local extensions used to track with Intra-Switch CDR. The number of tracked extensions can vary by system.
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Chapter 66: IP Address Mapping
Defines feature characteristics that depend on the IP address. Example command: change ip-network-map
Emergency Location Extension
The emergency location extension for this station. Accepts up to sixteen digits. Allows the system to properly identify the location of a caller who dials a 911 emergency call from this station. An entry in this field must be of an extension type included in the dial plan, but does not have to be an extension on the local system. It can be a Uniform Dial Plan (UDP) extension. A blank entry is typically used for an IP softphone dialing in through PPP from somewhere outside the network. For administered emergency numbers, the feature functions as follows:
· If the emergency location extension administered on the station screen is the same as the emergency location extension administered on the IP Address Mapping screen, the feature sends the station's own extension to the Public Safety Answering Point (PSAP).
· If the emergency location extension for the station is different from the emergency location extension administered here, the feature sends the extension administered here to the PSAP. Caution: On the ARS Digit Analysis Table, administer 911 to be call type emer or alrt in order for the E911 Emergency feature to work properly.
Related links Emergency Location Ext on page 770
From IP Address
The starting IP address. IPv6 format is supported.
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IP Address Mapping
Network Region
Valid Entry 1 to 2000
Usage
The network region for the IP address range. For SIP, the value for this field must correlate with the configured network region for this range of addresses. This field must contain a non-blank value if the corresponding From IP Address contains a non-blank value.
Subnet Bits
Valid Entry 0 to 64 blank
Usage
The number of bits of the subnet mask. Used in conjunction with the From IP Address to specify the end of the ID address range if a To IP Address is not administered.
To IP Address
The terminating IP address. IPv6 format is supported.
VLAN
Sends VLAN instructions to IP endpoints such as IP telephones and softphones. This field does not send VLAN instructions to the PROCR (S8300E Server), CLAN, and Media Processor boards.
Valid Entry 0 to 4094 n
Usage The virtual LAN value. Disabled
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Chapter 67: IP Codec Set
Use this screen to specify the type of codec used for voice encoding and companding (compression/ decompression). The default codec is set for G711. The G711 provides the highest voice quality, but G711 uses the maximum bandwidth. If G711 does not meet the required voice-quality or bandwidth trade-off specification, you can change the G711 default setting. To modify the values of the fields on this screen, use the following command: change ip-codec-set n, where n is the IP codec set number.
Note: The command change ip-media-parameters n works similar to the command change ip-codec-set n. Both the commands are valid.
IP Codec Set: page 1
Defines the allowed codecs and packet sizes used between VoIP resources. Enables silence suppression on a per-codec basis and dynamically displays the packet size in milliseconds for each codec in the set, based on the number of frames administered per packet.
Audio Codec
Specifies the audio codec used for this codec set. · G.711A (a-law) · G.711MU (mu-law) · G.722-64k · G.722.1-24k · G.722.1-32k · G.722.2 · G.723-5.3 · G.723-6.3
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IP Codec Set
· G.726A-32K · G.729 · G.729A · G.729B · G.729AB · SIREN14-24k · SIREN14-32k · SIREN14-48k · SIREN14-S48k · SIREN14-S56k · SIREN14-S64k · SIREN14-S96k · OPUS-NB12K · OPUS-NB16K · OPUS-WB20K · OPUS-SWB24K
Important:
Include at least two codecs for every telephone to avoid incompatible codecs. Only one OPUS codec is allowed on ip-codec form. Use the codecs specified in the following table for the telephones shown.
Telephone All Avaya IP Telephones 4601 4602 4602SW 4620SW 4621SW 4622SW
Codec to use G.711, G.729B
add G.726A (requires firmware R2.2)
Codec Set
The number assigned to this Codec Set.
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IP Codec Set: page 1
Encrypted SRTCP
Use the Encrypted SRTCP field to select a policy option to activate SRTCP for all call connections within IP network regions governed by this IP codec set. This field works only if the associated RTP channel is operating in encrypted mode, that is, srtcp-aes128 or srtcp-aes256. The system displays this field only when you enable the Media Encryption Over IP field on the change system-parameters customer-options screen.
Valid entry enforce-enc-srtcp best-effort
enforce-unenc-srtcp
Usage
To permit only encrypted SRTCP calls. Use this option to achieve high security standards.
If you want Communication Manager to facilitate negotiation of the encrypted SRTCP parameter on call connections that traverse the IP network regions that are governed by this IP codec set.
To permit only calls that do not use media encryption. For networks that have Communication Manager Release 7.0 with older endpoints. This option is the default value for this field.
Frames Per Pkt
Specifies the number of frames per packet up to a packet size of 60 milliseconds (ms).
Valid Entry 1 to 6
Usage Default frame sizes for codecs: · G.711 and G.729: 2 frames (20 ms) · G.723: 1 frame (30 ms) · G.726A: 2 frames (20 ms) · OPUS-NB12K: 1 frame (20 ms) · OPUS-NB16K: 1 frame (20 ms) · OPUS-WB20K: 1 frame (20 ms) · OPUS-SWB24K: 1 frame (20 ms)
Media Encryption
This field specifies a priority listing of five options for the negotiation of encryption. Communication Manager provides bearer encryption according to the administered priority order. The selected option for an IP codec set applies to all codecs defined in that set. The ip-codec-set screen displays the Media Encryption field only if you enable the Media Encryption over IP field.
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IP Codec Set
Note: SRTP is a media encryption standard defined in RFC 3711 as a profile of RTP. Communication Manager supports the following functionality as specified in RFC 3711.
· Encryption of RTP (optional but recommended) · Authentication of RTCP streams (mandatory) · Authentication of RTP streams (optional but recommended) · Protection against replay
Note: AES and AEA are Avaya proprietary techniques and not recommended for use. The SRTCP options are recommended. The following four choices are the recommended options for use:
· 10-srtp-aescm256-hmac80 · 11-srtp-aescm256-hmac32 · 1-srtp-aescm128-hmac80 · 2-srtp-aescm128-hmac32
Valid entry aes
aea
1-srtp-aescm128hmac80 2-srtp-aescm128hmac32 3-srtp-aescm128hmac80-unauth 4-srtp-aescm128hmac32-unauth 5-srtp-aescm128hmac80-unenc
Usage Advanced Encryption Standard (AES), a standard cryptographic algorithm for U.S. government organizations to protect sensitive (unclassified) information. Use this option to encrypt the following links: · Server-to-gateway (H.248) · Gateway-to-endpoint (H.323) Avaya Encryption Algorithm. Use this option as an alternative to AES encryption when: · All endpoints within a network region using this codec set must be encrypted. · All endpoints communicating between two network regions and administered to
use this codec set must be encrypted. 1-Encrypted/Authenticated RTP with 80-bit authentication tag
2-Encrypted/Authenticated RTP with 32-bit authentication tag
3-Encrypted RTP but not authenticated
4-Encrypted RTP but not authenticated
5-Authenticated RTP with 80-bit authentication tag but not encrypted
Table continues...
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IP Codec Set: page 2
Valid entry
Usage
6-srtp-aescm128- 6-Authenticated RTP with 32-bit authentication tag but not encrypted hmac32-unenc
7-srtp-aescm128hmac80-unencunauth
7-Unencrypted/Unauthenticated RTP
8-srtp-aescm128hmac32-unencunauth
8-Unencrypted/Unauthenticated RTP
Note: For stations, the only value supported is srtp-aescm128-hmac80. H.323 IP trunks support all eight listed algorithms.
10-srtpaescm256hmac80
10-AES-256 Encrypted/Authenticated RTP with 80-bit authentication tag
11-srtp-aescm256- 11-AES-256 Encrypted/Authenticated RTP with 32-bit authentication tag hmac32
none
Unencrypted media stream. This entry is the default.
Packet Size (ms)
The packet size in milliseconds.
Silence Suppression
Enables or disables RTP-level silence suppression on the audio stream.
IP Codec Set: page 2
You can configure the following capabilities on the codec set: · Enable or disable Direct-IP Multimedia for videophone transmissions. · Enable or disable endpoints in the assigned network region to send fax, modem, or TTY calls over IP trunks. · Define the mode that the system uses to route the fax, modem, or TTY calls. · Determine whether redundancy must be added to the transmission for higher reliability and quality.
You must assign these capabilities to the codec set, and assign the codec set to a network region for endpoints in that region to be able to use the capabilities established on this screen.
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IP Codec Set
Using V.150.1, you can configure fax and modem so that both use a relay mode together in an IP codec set. You can set the FAX Mode field to t.38standard for fax calls to use t.38. However, modem calls use V.150.1.
Note: Transporting modem signals over IP between Avaya Communication Manager systems and media gateways is a proprietary implementation. The selection of the V.150.1 Modem Relay codec choice is not proprietary. For modem transport interoperability with third-party gateways, v150mr is the recommended mode of operation. Caution: If users are using Super G3 fax machines and modems, do not assign these fax machines to a network region with an IP codec set that is modem-enabled as well as fax-enabled. If its Codec set is enabled for both modem and fax signaling, a Super G3 fax machine incorrectly tries to use the modem transmission instead of the fax transmission. Therefore, assign modem endpoints to a network region that uses a modem-enabled IP Codec set, and the Super G3 fax machines to a network region that uses a fax-enabled IP Codec set.
Allow Direct-IP Multimedia
To enable or disable direct multimedia, use the following codecs: · H.261 · H.263 · H.264 (video) · H.224 · H.224.1 (data, far-end camera control).
Clear-channel Mode
Enables or disables supporting this codec set for BRI data calls.
ECM
With Error Correction Mode (ECM), fax endpoints correct the fax transmission error without the need to retransmit multiple pages. The ECM field is available only when FAX Mode is t.38-standard or t.38-G711-fallback.
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Valid entry y
n
IP Codec Set: page 2
Usage ECM-capable gateways use the T.38 protocol to relay T.30 ECM signaling and ECM frames between a local Group 3 fax machine and a far-end T.38 receiver. The default value is y. ECM-capable gateways do not use the T.38 protocol to relay T.30 ECM signaling and ECM frames.
FAX Mode
Valid entry off relay pass-through T.38-standard
T.38-G711-fallback
Usage
For users using this mode, Communication Manager treats a fax call as an ordinary voice call.
For users using this mode, Communication Manager uses the Avaya proprietary relay mode for fax transmissions over IP network facilities only between Avaya devices. This mode can be coupled with Redundancy.
For users using this mode, Communication Manager uses the Avaya proprietary pass-through mode for in-band fax transmissions over IP network facilities only between Avaya devices. This mode can be coupled with Redundancy.
For users using this mode, Communication Manager uses the T.38 protocol for fax transmission over IP network facilities.
Communication Manager supports the transition of an existing audio call to a fax call, for both H.323 and SIP calls.
For users using this mode, Communication Manager uses the T.38 protocol for fax transmission only if the protocol can be successfully negotiated with the peer SIP entity.
Otherwise, Communication Manager falls back to G.711 for fax transmission. This mode requires a G.711 codec to be administered on the ip-codec-set screen.
Note:
If you have a telephone on an IP trunk that is close to a fax machine, the telephone can change into the fax mode by picking up the tones from the fax machine. To prevent the telephone from changing into the fax mode, set the FAX mode to off and put the fax machines in an Automatic Route Selection (ARS) partition that uses only circuit switched trunks, even for intergateway fax calls.
Maximum Call Rate for Priority Direct-IP Multimedia
The system displays this field only if Allow Direct-IP Multimedia is set to y.
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IP Codec Set
Maximum Call Rate for Direct-IP Multimedia
Use this field to limit the size of bandwidth used for video calls. The system displays this field appears only if Allow Direct-IP Multimedia is set to y. The default value is 384 kilobits. You can enter a value between 64 kilobits and 15360 kilobits, with an increment of 64 kilobits. A typical video endpoint does not require more than 2048 kilobits.
Media Connection IP Address Type Preferences
Use this field to specify the type of IP address and the preference of the IP address type to be used for media. The following are the valid entries:
· IPv4/IPv6 · IPv6/IPv4 · IPv4/none · IPv6/none
Modem Mode
Valid entry off relay
pass-through v150mr
Usage
In the off mode, a modem call is treated as an ordinary voice call. The default mode for new installations and upgrades is off.
In the Avaya proprietary relay mode, Avaya devices detect the tones of the call and use the appropriate modulation protocol or TTY to terminate or originate the modem calls over IP network.
In the Avaya proprietary pass-through mode, Avaya devices detect the tones of the modem call and use G.711 encoding to process the call over the IP network.
In the v150mr mode, Avaya devices detect the tones of the modem call and use V.150.1 protocol to transmit V-series modem signals between secure terminals and third-party SIP gateways.
Note:
V.150.1 protocol is a standard recommended by International Telecommunication Union (ITU) to use a modem over IP networks that support dialup modem calls.
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IP Codec Set: page 2
Packet Size
Note: This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers. The Packet Size column supports choices of 10, 20, 30, 40, 50, or 60 millisecond IP packet size to be used in the RFC 4040 transport. The default choice is 20 milliseconds. If the packet size is of value other than 20 milliseconds, then the SDP Offer must include an Attribute format string. For example, for an administered Packet Size of 10 milliseconds, Communication Manager sends an SDP Offer message that includes the attribute string of a=ptime 10.
Redundancy
Use this field to assign the number of duplicate or redundant packets that must be sent in addition to the primary packet. You can enter a value from 1 to 3. The default value is 0.
Note: For the pass-through and the clear-channel modes, you can use the default value or assign 1. This field does not apply to the v150mr mode. If you set the Modem Mode field to v150mr, the system sets the Redundancy field to display-only with the default value. The Redundancy field supports choices of 0 and 1 for the SIP64K field. 0 designates that no redundant media packets are transmitted. 1 designates that a single redundant packet should be transmitted for each media stream packet that is sent.
Note: This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
SIP 64K Data
With SIP 64K Data, Communication Manager controls the mechanism to enable the support of the RFC 4040 media service.
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IP Codec Set
Valid Entry y
n
Usage
Communication Manager searches for an RFC 4040 capable gateway, G450 or G430. If the given gateway does not support the RFC 4040 feature, Communication Manager searches for an available gateway in the connected network region. If Communication Manager cannot detect a suitable gateway within the network region, the call request is dropped.
The default value is n.
Note:
The G650 gateway does not support the RFC 4040 feature. G430 and G450 gateways support the RFC 4040 feature. The gateway in the network region must support the RFC 4040 feature for Communication Manager to connect RFC 4040 calls.
Communication Manager uses RFC 4040 Clear Mode data transport to support the media transport of ISDN traffic. The ISDN traffic is directed to a destination that is reached through a SIP trunk.
Associated with the SIP 64K Data field are two other fields, Redundancy and Packet Size (ms). Communication Manager communicates the values of the Redundancy and Packet Size (ms) fields to the media gateway so that the gateway properly operates with the DSP conversion of the TDM media into an IP media stream.
TDD/TTY Mode
Valid entry off
US
UK pass-through
Usage
Turn off special TTY handling when using this codec set. In this case, the TTY transmission is treated like an ordinary voice call. With a codec set that uses G.711, this setting must send TTY calls to non-Avaya systems. However, there might be errors in character transmissions.
For users in regions using this codec, use U.S. Baudot 45.45 mode for TTY transmissions over IP network facilities. This is the default value for new installations and upgrades.
For users in regions using this codec, use U.K. Baudot 50 mode for TTY transmissions over IP network facilities.
For users in regions using this codec, use the Avaya proprietary pass-through mode for TTY transmissions over IP network facilities.
IP Codec Set: page 3
The system displays this page only when you set the Modem Mode field to v150mr.
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IP Codec Set: page 3
Modem Relay Preferred (NoAudio)
If you set this field to y, Communication Manager includes the NoAudio codec in the SIP SDP offer/answer message. Enable this field when a gateway interoperates with the devices that support offer/answer NoAudio codec. The default value is n.
Modulation Mode
Avaya V.150.1 supports the V.32, the V.34, the V.90, and the V.92 modem modulations. For users who need other modulations, you must set the Modem Mode field to pass-through. The following fields show whether V.150.1 is enabled to transport the respective modulations as Modem Relay:
V.32
Valid entry y n
Usage Use this entry to enable V.32 modem modulation. The default value is y. Use this entry to disable V.32 modem modulation.
V.34
Valid entry y n
Usage Use this entry to enable V.34 modem modulation. The default value is y. Use this entry to disable V.34 modem modulation.
V.90
Valid entry y
n
Usage Use this entry to enable the V.90 digital modem modulation. The default value is y.
Note:
The system sets the V.34 field to display-only with the default value y. Use this entry to disable the V.90 digital modem modulation.
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IP Codec Set
V.92
Valid entry y
n
Usage Use this entry to enable the V.92 digital modem modulation.
Note: The system sets the V.34 field to display-only with the default value y. Use this entry to disable the V.92 digital modem modulation. The default value is n.
SPRT Retransmissions
Use this field to assign the maximum number of data retransmissions to Transport Channel 1 (TC1) and Transport Channel 2 (TC2). You can assign a value from 1 to 32. The default value is 3.
SPRT TC0 Payload Size
Use this field to assign the maximum payload size to SPRT Transport Channel 0. You can enter a value from 140 to 256. The default value is 140 bytes.
Note: Maximum SPRT payload sizes might vary across different gateways depending on the firmware and the hardware version.
SPRT TC1 Payload Size
Use this field to assign the maximum payload size for SPRT Transport Channel 1. You can enter a value between 132 to 256. The default value is 132 bytes.
Note: Maximum SPRT payload sizes might vary across different gateways depending on the firmware and the hardware versions.
SPRT TC1 Window Size
Use this field to assign the window size to SPRT Transport Channel 1. You can enter a value from 32 to 96. The default value is 32 bytes.
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IP Codec Set: page 3
SPRT TC2 Payload Size
Use this field to assign the maximum payload size to SPRT Transport Channel 2. You can enter a value from 132 to 256. The default value is 132 bytes.
Note: Maximum SPRT payload sizes might vary across different gateways depending on the firmware and the hardware versions.
SPRT TC2 Window Size
Use this field to assign the window size to SPRT Transport Channel 2. You can enter a value from 8 to 32. The default value is 8 bytes.
SPRT TC3 Payload Size
Use this field to assign the maximum payload size to SPRT Transport Channel 3. You can enter a value from 140 to 256. The default value is 140 bytes.
Note: Maximum SPRT payload sizes might vary across different gateways depending on the firmware and the hardware versions.
SPRT Timer TC1-TA01
Use this field to set the value of the acknowledgement timer of Transport Channel 1. This field accepts values between 50 and 1000 but only in multiples of 10. The default value is 90.
SPRT Timer TC1-TA02
Use this field to set the value of the acknowledgement update timer of Transport Channel 2. This field accepts values between 50 and 1000 but only in multiples of 10. The default value is 130.
SPRT Timer TC1-TR03
Use this field to set the value of the retransmit timer of Transport Channel 1. This field accepts values between 50 and 1000 but only in multiples of 10. The default value is 500.
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IP Codec Set
SPRT Timer TC2-TA01
Use this field to set the value of the acknowledgement timer of Transport Channel 2. This field accepts values between 50 and 1000 but only in multiples of 10. The default value is 90.
SPRT Timer TC2-TA02
Use this field to set the value of the acknowledgement update timer of Transport Channel 2. This field accepts values between 50 and 1000 but only in multiples of 10. The default value is 500.
SPRT Timer TC2-TR03
Use this field to set the value of the retransmit timer of Transport Channel 2. This field accepts values between 50 and 1000 but only in multiples of 10. The default value is 500.
SSE Inter-Packet Interval
Use this field to assign an interval between the State Signaling Events (SSE) packets. You can enter a value between 10 to 40 with the increments of 10. The default value is 20 mSec.
SSE Repetition Redundancy (Packets)
Use this field to assign the number of redundant SSE packets for transmission. The redundancy option provides robust transport of data packets across an IP-based network. You can enter a value from 0 to 3. The default value is 3.
V.42 Error Correction
V.42 is an error correction protocol for V.150.1 media transport between modems. If you set this field to y, the gateways use the V.42 Error Correction protocol. The default value is n.
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Chapter 68: IP Interfaces
Assigns a network region to an IP interface device, or administers Ethernet options. The appearance of the IP Interfaces screen can vary according to the interface type you are administering, and your system's configuration.
Note: After starting the process of administering the IP interface for the TN2602AP circuit pack, any active calls continue to use the TN2602AP circuit pack's physical IP address for the connection, not the virtual IP address administered here. Therefore, any calls that continue after administering the virtual address, drop in the event of an interchange. Example command: add ip-interface n, where n is the board location.
IP Interface: page 1
Allow H.248 Gateways
Controls whether branch gateways (G700, G450, G430) can or cannot register on the interface.
Valid Entry y
n
Usage
On a simplex main server, enables H.248 endpoint connectivity to the Processor Ethernet (PE) interface. Used for a Survivable Remote Server (Local Survivable Processor).
Disables H.248 endpoint connectivity to the PE interface. H.248 endpoint connectivity using the PE interface on a Survivable Core Server (Enterprise Survivable Server) is not supported.
Allow H.323 Endpoints
Controls whether H.323 endpoints can or cannot register on the interface.
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IP Interfaces
Valid Entry y
n
Usage
On a simplex main server, enables H.323 endpoint connectivity to the Processor Ethernet (PE) interface. Used for a Survivable Remote Server (Local Survivable Processor).
Disables H.323 endpoint connectivity to the PE interface. H.323 endpoint connectivity using the PE interface on a Survivable Core Server (Enterprise Survivable Server) is not supported.
Code/Sfx
Circuit pack TN code and suffix.
Note: The 4606, 4612, and 4624 telephones do not support the bearer duplication feature of the TN2602AP circuit pack. If these telephones are used while an interchange from active to standby media processor is in process, calls might be dropped. Related links Critical Reliable Bearer on page 450
Critical Reliable Bearer
Enables or disables a duplicate TN2602 circuit pack in a port. Available only with the TN2602.
Note: The 4606, 4612, and 4624 telephones do not support the bearer duplication feature of the TN2602AP circuit pack. If these telephones are used while an interchange from active to standby media processor is in process, calls might be dropped.
Enable Ethernet Interface
Valid Entry y n
Usage
Enables the Ethernet port associated with the TN2602AP circuit pack.
Disables when there is no standby, or when the standby has been disabled. Should be disabled before administering the IP interface.
Ethernet Link
The administered link number for an Ethernet link.
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IP Interface: page 1
Gateway Node Name
The gateway node name associated with the IP address of the LAN gateway associated with the TN2602AP. This entry also applies to the second TN2602AP circuit pack when Critical Reliable Bearer is enabled. Accepts up to 15 characters.
Note:
The 4606, 4612, and 4624 telephones do not support the bearer duplication feature of the TN2602AP circuit pack. If these telephones are used while an interchange from active to standby media processor is in process, calls might be dropped.
Related links Critical Reliable Bearer on page 450
Gatekeeper Priority
Available only if H.323 endpoints are enabled and the Communication Manager server is a main server or a Survivable Remote Server. Unavailable on a Survivable Core Server.
Valid Entry 1 to 9
Usage
Sets a priority on the interface that affects where the interface is displayed on the gatekeeper list. The value in this field is used on the alternate gatekeeper list. The lower the number, the higher the priority. Default is 5.
Related links Allow H.323 Endpoints on page 449
Network Region
Valid Entry 1 to 2000
Usage
The value of the network region where the TN2602AP resides. This entry also applies to the second TN2602AP circuit pack when Critical Reliable Bearer is enabled.
Note:
The 4606, 4612, and 4624 telephones do not support the bearer duplication feature of the TN2602AP circuit pack. If these telephones are used while an interchange from active to standby media processor is in process, calls might be dropped.
Related links Critical Reliable Bearer on page 450
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IP Interfaces
Network uses 1's for Broadnet Addresses
Allows or denies using a broadcast address to send the same message to all systems or clients on a local area network.
Node Name
The node name associated with the IP address of the TN2602AP circuit pack. Accepts up to 15 characters.
Related links Near-end Node Name on page 714 Name on page 475 Group Type on page 708
Receive Buffer TCP Window Size
Valid Entry 512 to 8320
Usage
The number of bytes allotted for the buffer that receives TCP data for a TN799 (CLAN) circuit pack. The default is 512.
Slot
The slot location. Requires entry of the location of the second TN2602AP circuit pack for a nonduplicated board.
Note:
The 4606, 4612, and 4624 telephones do not support the bearer duplication feature of the TN2602AP circuit pack. If these telephones are used while an interchange from active to standby media processor is in process, calls might be dropped.
Valid Entry 1 to 64 A to E 0 to 20
Usage First and second characters are the cabinet number. Third character is the carrier. Fourth and fifth character are the slot number.
Related links Critical Reliable Bearer on page 450
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IP Interface: page 1
Subnet Mask
A 32-bit binary number that divides the network ID and the host ID in an IP address. This is the subnet mask for TN2602AP. Also applies to the second TN2602AP circuit pack when the critical reliable bearer is enabled.
Related links Critical Reliable Bearer on page 450
Target socket load
The maximum number of sockets targeted for this interface. The default is 80% of the platform maximum. Used for load balancing endpoint traffic across multiple IP interfaces. Controls the percentage of sockets allocated to each IP interface within the same Gatekeeper Priority. When all the IP interfaces within the same Gatekeeper Priority exceeds the target number allocated, the system continues to add sockets until the interface is at its maximum capacity. Available only with a procr type IP interface.
Note:
The 4606, 4612, and 4624 telephones do not support the load balancing feature of the TN2602AP circuit pack.
Valid Entry 1 to 3500
Usage HP DL360p G8, HP DL360 G9, Dell R620, and Dell R630, and duplex servers
Related links Type on page 454
Target socket load and Warning level
Controls the percentage of sockets allocated to each IP interface within the same Gatekeeper Priority. When all the IP interfaces within the same Gatekeeper Priority exceeds the target number allocated, the system continues to add sockets until the interface is at its maximum capacity. If the targeted percentage is exceeded on a CLAN, a warning alarm is generated.
If there is only one IP interface within a priority, the target socket load and warning level is not used for load balancing. A value in this field can be used to receive an error or a warning alarm if the targeted value is exceeded. Available only with CLAN type IP interfaces.
Note:
The 4606, 4612, and 4624 telephones do not support the load balancing feature of the TN2602AP circuit pack.
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IP Interfaces
Valid Entry 1 to 499
Usage
The maximum number of sockets targeted for this interface. If the number of sockets exceeds the targeted number, a warning alarm is generated. The default is 400.
Related links Type on page 454
Type
The type of IP interface.
Valid Entry clan VAL medpro procr
Usage Control Local Area Network (C-LAN) board Voice Announcement LAN board Media Processor board Processor -- S8300E and duplex media servers
VLAN
Sends Virtual Local Area Network (VLAN) instructions to the PROCR (8300E/Duplex Media Servers), CLAN, and Media Processor boards. It does not send VLAN instructions to IP endpoints such as IP telephones and softphones. Unavailable for Voice Announcement over LAN (VAL) boards.
Valid Entry 0 to 4095 n
Usage Specifies the virtual LAN value. Disabled. This is the default.
IP Interface: page 2
ETHERNET OPTIONS
The Ethernet port associated with the TN2602AP must be disabled before any changes can be made to these fields.
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IP Interface: page 2
Auto?
Valid Entry y n
Usage Enables auto-negotiation. Default. Applies manual speed and duplex settings.
Duplex
Available only if auto-negotiation is disabled.
Valid Entry Full Half
Usage The full duplex setting for this IP board. Default when speed is set to 100 Mbps. The half duplex setting for this IP board. Default.
Related links Auto? on page 455
Speed
Available only if auto-negotiation is disabled.
Valid Entry 10Mbps 100Mbps
Usage The speed of the Ethernet connection.
The only speed option available for the TN2602AP circuit pack is 100Mbps. This is the default and cannot be changed.
Related links Auto? on page 455
IPV6 PARAMETERS
Enable Ethernet Interface
Valid Entry y n
Usage
Enables the Ethernet port associated with the TN2602AP circuit pack.
Disables when there is no standby, or when the standby has been disabled. Should be disabled before administering the IP interface.
Ethernet Link
The administered link number for an Ethernet link.
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IP Interfaces
Gateway Node Name
The gateway node name associated with the IP address of the LAN gateway associated with the TN2602AP. This entry also applies to the second TN2602AP circuit pack when Critical Reliable Bearer is enabled. Accepts up to 15 characters.
Note:
The 4606, 4612, and 4624 telephones do not support the bearer duplication feature of the TN2602AP circuit pack. If these telephones are used while an interchange from active to standby media processor is in process, calls might be dropped.
Related links Critical Reliable Bearer on page 450
Node Name
The node name associated with the IP address of the TN2602AP circuit pack. Accepts up to 15 characters.
Related links Near-end Node Name on page 714 Name on page 475 Group Type on page 708
Subnet Mask
A 64-bit binary number that divides the network ID and the host ID in an IP address. This is the subnet mask for TN2602AP. Also applies to the second TN2602AP circuit pack when the critical reliable bearer is enabled.
Note:
The 4606, 4612, and 4624 telephones do not support the bearer duplication feature of the TN2602AP circuit pack. If these telephones are used while an interchange from active to standby media processor is in process, calls might be dropped.
Related links Critical Reliable Bearer on page 450
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IP Interfaces: page 3
IP Interfaces: page 3
VOIP/NETWORK THRESHOLDS
Enable VoIP/Network Thresholds
Enables or disables the recording of Voice/Network Statistics at a system level for a single media processor board. This applies to both TN2602 boards, if duplicated. Any changes to the value of this field, results in an updated message sent to the media processor board.
Jitter (ms)
Available only if VoIP/Network thresholds are enabled and the board type is a media processor.
Valid Entry 0 to 9999
Usage
The unacceptable jitter coming into the media processor board at which point data is captured and sent to Communication Manager. Default is 50 milliseconds.
Related links Enable VoIP/Network Thresholds on page 457 Type on page 454
Packet loss (%)
Available only if VoIP/Network thresholds are enabled and the board type is a media processor.
Valid Entry 0 to 100
Usage The percentage of the unacceptable packet loss coming into the administered media processor board. Default is 5.
Note:
xxx indicates 100% packet loss.
Related links Enable VoIP/Network Thresholds on page 457 Type on page 454
RT Delay (ms)
Available only if VoIP/Network thresholds are enabled and the board type is a media processor.
Valid Entry 1 to 9999
Usage
Round Trip Delay is the unacceptable elapsed time for a packet to reach a remote location and return. Default is 500 milliseconds.
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IP Interfaces
Related links Enable VoIP/Network Thresholds on page 457 Type on page 454
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Chapter 69: IP network region
Configures within-region and between-region connectivity settings for all VoIP resources and endpoints within a given IP region. The first page is used to modify the audio and QoS settings. The Codec Set field on this page reflects the CODEC set that must be used for connections between telephones within this region or between telephones and MedPro/Prowler boards and gateways within this region. The ability to do NAT shuffling for direct IP-to-IP audio connections is also supported. Example command: change ip-network-region n, where n is the network region number.
IP network region: page 1
NR Group
Use this field to assign a network region group to the network region. You can enter a value from: · 1 to 2000 for large systems. · 1 to 250 for small systems.
Do not leave the field blank. You can assign multiple network regions to the same network region group. During translations upgrade, the system assigns a network region group to each network region. For example, network region 45 will be assigned to network region group 45. The network region group names will also be assigned. If the network region has a name, the name is used for the associated network region group name. If the network region name is blank, a unique network region group name is created by appending the network region group number to the characters NRG. For example, NRG 16.
Authoritative Domain
The domain for which this network region is responsible. The system displays the Authoritative Domain in the From header of any SIP messages. Accepts a name or IP address consisting of up to 20 characters.
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IP network region
Location
Note:
If the Multinational Locations feature is enabled, and IP telephones derive their network region from the IP Network Map, administer this field with a valid value (1 to 2000). With this, the IP endpoints can use the right VoIP resources.
Valid Entry
1 to 250 or 1 to 2000
Depending on the server configuration. For more information, see Avaya Aura® Communication Manager System Capacities Table.
blank
Usage
Assigns the location number to the IP network region. Correct date and time information, and trunk routing based on IP network region can be configured. The IP endpoint uses this as its location number. See the Location sections in Avaya Aura® Communication Manager Feature Description and Implementation, for the other ways to assign a location, and for a list of features that use location.
Obtains the location from the cabinet containing the CLAN or the gateway where the endpoint is registered. By default, the value is blank.
Name
A description of the region. Accepts up to 20 characters.
Region
The number of the network region being administered.
Stub Network Region
Using this field, you can create a core network region or a stub network region.
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Valid entry y
n
IP network region: page 1
Usage
Creates a stub network region. In the dst rgn field, you must enter the number of the destination network region that directly connects with this stub network region.
Note:
If you are converting a core network region to a stub network region, you must ensure that the core network region is directly connected with only one core network region. A stub network must have only one direct connection with a core network region.
Creates a core network region. For network regions 1 to 250, the default value of this field is n. For network regions 251 to 2000, this field is noneditable and displays a default of y.
MEDIA PARAMETERS
Codec Set
Valid Entry 1 to 7
Usage The number for the codec set for the region.
Intra-region IP-IP Direct Audio
Allows direct audio connections between IP endpoints within a network region.
Valid Entry y n native(NAT)
translated(NAT)
Usage
Saves on bandwidth resources and improves sound quality of voice over IP transmissions.
Might be used if, for example, the IP telephones within the region are behind two or more firewalls.
The IP address from which audio is to be received for direct IP-to-IP connections within the region is that of the telephone/ softphone itself (without being translated by NAT). IP telephones must be configured behind a NAT device before this entry is enabled.
The IP address from which audio is to be received for direct IP-to-IP connections within the region is to be the one with which a NAT device replaces the native address. IP telephones must be configured behind a NAT device before this entry is enabled.
Inter-region IP-IP Direct Audio
Allows direct audio connections between IP endpoints in different regions
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Valid Entry y n native(NAT)
translated(NAT)
Usage
Saves on bandwidth resources and improves sound quality of voice over IP transmissions.
Might be used if, for example, the IP telephones within the region are behind two or more firewalls.
The IP address from which audio is to be received for direct IP-to-IP connections between regions is that of the telephone itself (without being translated by NAT). IP telephones must be configured behind a NAT device before this entry is enabled.
The IP address from which audio is to be received for direct IP-to-IP connections between regions is to be the one with which a NAT device replaces the native address. IP telephones must be configured behind a NAT device before this entry is enabled.
IP Audio Hairpinning
If enabled, IP endpoints connect through the IP circuit pack in the server in IP format to bypass the Communication Manager TDM bus.
UDP PORT RANGE
UDP Port Range Min
Valid Entry 1024 to 65534
Usage
The minimum range of the UDP port number used for audio transport. Defaults to 2048 to 3028.
UDP Port Range Max
Valid Entry 1025 to 65535
Usage
The maximum range of the UDP port number used for audio transport. Defaults to 2048 to 3028.
DIFFSERV/TOS PARAMETERS
Audio PHB Value
Provides scalable service discrimination in the Internet without per-flow state and signaling at every hop.
Valid Entry 0 to 63
Usage The decimal equivalent of the DiffServ Audio PHB value. Default is 46.
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IP network region: page 1
Call Control PHB Value
Provides scalable service discrimination in the Internet without per-flow state and signaling at every hop.
Valid Entry 0 to 63
Usage The decimal equivalent of the Call Control PHB value. Default is 34.
Video PHB Value
Valid Entry 0 to 63
Usage The decimal equivalent of the DiffServ Video PHB value. Default is 26.
802.1P/Q PARAMETERS
Audio 802.1p Priority
Valid Entry 0 to 7
Usage Provides Layer 2 priority for Layer 2 switches. Changes take effect after circuit pack reset, telephone reboot, or system reset.
Call Control 802.1p Priority
Valid Entry 0 to 7
Usage Provides Layer 2 priority for Layer 2 switches. Changes take effect after circuit pack reset, telephone reboot, or system reset.
Video 802.1p Priority
Valid Entry 0 to 7
Usage
The Video 802.1p priority value. Changes take effect after circuit pack reset, telephone reboot, or system reset.
AUDIO RESOURCE RESERVATION PARAMETERS
Retry upon RSVP Failure Enabled
Enables or disables retries when RSVP fails. Available only if RSVP is enabled. Related links RSVP Enabled on page 464
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IP network region
RSVP Enabled
Enables or disables RSVP.
RSVP Profile
Available only if RSVP is enabled. Set this field to what you have configured on your network.
Valid Entry guaranteed-service
controlled-load
Usage
Limits end-to-end queuing delay from sender to receiver. This setting is best for VoIP applications.
This subset of guaranteed-service provides for a traffic specifier, but not end-to-end queuing delay.
RSVP Refresh Rate (secs)
Available only if RSVP is enabled. The system displays this field only if the RSVP Enabled field is set to y.
Valid Entry 1 to 99
Usage The RSVP refresh rate in seconds.
Related links RSVP Enabled on page 464
RSVP unreserved (BBE) PHB Value
Valid Entry 0 to 63
Usage
The BBE codepoint is used whenever an RSVP reservation is being obtained (pending), or has failed in some way, to provide efficient service to the voice stream.
H.323 IP ENDPOINTS
H.323 Link Bounce Recovery
Enables or disables the H.323 Link Bounce Recovery feature for this network region. The default is enabled.
Idle Traffic Interval (seconds)
Valid Entry 5 to 7200
Usage
The maximum traffic idle time in seconds after which a TCP Keep-Alive (KA) signal is sent from the endpoint.. Default is 20.
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IP network region: page 2
Keep-Alive Interval (seconds)
Valid Entry 1 to 120
Usage
Sets the interval between TCP Keep-Alive re-transmissions. When no ACK is received for all retry attempts, the local TCP stack ends the TCP session and the associated socket is closed. Default is 5.
Keep-Alive Count
Valid Entry 1 to 20
Usage
Sets the number of times the Keep-Alive message is transmitted if no ACK is received from the peer. Default is 5.
IP network region: page 2
ALTERNATIVE NETWORK ADDRESS TYPES
To control the call processing behavior to send Alternative Network Address Types (ANAT) offer for the IP network region.
ANAT Enabled
Use this field to control the call processing behavior of the Alternative Network Address Types (ANAT) for the IP network region.
Valid entry y
n
Usage
Communication Manager sends ANAT offer to SIP elements in the IP network region when you set this field to y as well as system-wide ANAT Enabled field to y.
Communication Manager does sends ANAT offer to SIP elements in the IP network region.
RTCP Reporting to Monitor Server Enabled
If enabled, sends RTCP Reports to a special server, such as for the VMON tool. Note:
Regardless of how this field is administered, RTCP packets are always sent peer-to-peer.
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IP network region
RTCP MONITOR SERVER PARAMETERS
IPV4 Server Port
Available only if RTCP Reporting is enabled and if Default Server Parameters are disabled.
Valid Entry 1 to 65535
Usage The port for the RTCP Monitor server. Default is 5005.
IPV6 Server Port
Available only if RTCP Reporting is enabled and if Default Server Parameters are disabled.
Valid Entry 1 to 65535
Usage The port for the RTCP Monitor server. Default is 5005.
RTCP Report Period (secs)
Available only if RTCP Reporting is enabled and if Default Server Parameters are disabled.
Valid Entry 5 to 30
Usage The report period for the RTCP Monitor server in seconds.
Related links Use Default Server Parameters on page 466
Server IPV4 Address
The IPv4 address for the RTCP Monitor server. Available only if RTCP Reporting is enabled and if Default Server Parameters are disabled.
Server IPV6 Address
The IPv6 address for the RTCP Monitor server. Available only if RTCP Reporting is enabled and if Default Server Parameters are disabled.
Use Default Server Parameters
If enabled, uses the system-wide default RTCP Monitor server parameters. Available only if RTCP Reporting is enabled.
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IP network region: page 3
IP network region: page 3
This page covers the information for Inter-Gatewày Alternate Routing (IGAR), backup server names in priority order, and security procedures.
INTER-GATEWAY ALTERNATE ROUTING/DIAL PLAN TRANSPARENCY
If Inter-Gateway Alternate Routing (IGAR) is enabled for any row on subsequent pages, the following fields for each network region must be administered to route the bearer portion of an IGAR call.
Conversion to Full Public Number - Delete
Valid Entry 0 to 7
Usage The digits to delete
Conversion to Full Public Number - Insert
Valid Entry 0 to 16 blank
Usage
The number of digits to insert. International numbers should begin with plus (+). The Inter-Gateway Alternate Routing (IGAR) and Dial Plan Transparency (DPT) features convert the plus (+) digit to appropriate international access code when starting the trunk call.
Note:
The optional plus (+) at the beginning of the inserted digits is an international convention indicating that the local international access code must be dialed before the number.
Dial Plan Transparency in Survivable Mode
Valid Entry y
n
Usage
Enables the Dial Plan Transparency feature when a gateway registers with a Survivable Remote Server (Local survivable processor), or when a port network registers with a Survivable Core Server (Enterprise Survivable Server).
Default is n.
Incoming LDN Extension
An extension used to assign an unused Listed Directory Number for incoming IGAR calls.
Maximum Number of Trunks to Use for IGAR
It is necessary to impose a limit on the trunk usage in a particular port network in a network region when Inter-Gateway Alternate Routing (IGAR) is active. The limit is required because if there is a
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IP network region
major IP WAN network failure, it is possible to use all trunks in the network region(s) for IGAR calls.
Valid Entry 1 to 999, or blank
Usage
The maximum number of trunks to be used for Inter-gateway alternate routing (IGAR).
BACKUP SERVERS IN PRIORITY ORDER
Lists the backup server names in priority order. Backup server names should include Survivable Remote Server names and Survivable Core Server names. If you are using the Processor Ethernet, the backup servers list must include the survivable core PE address else the phones will not register to the survivable core during a failure. Any valid node name is a valid entry. Valid node names can include names of Customer LANs, ICCs, Survivable Core Servers, and Survivable Remote Servers.
H.323 SECURITY PROFILES
Permitted security profiles for endpoint registration in the network region. You must enter at least one security profile. Otherwise, no endpoint will be permitted to register from the region.
Valid Entry challenge
pin-eke strong
any-auth H323TLS
Usage
Includes the various methods of PIN-based challenge and response schemes in current use. This is a relatively weak security profile.
The H.235 Annex H SP1.
Permits the use of any strong security profile.
The H323TLS profile is the strongest security profile in Communication Manager.
Includes any of the security profiles.
Communication Manager apples this security profile when the network region of an H.323 phone is administered with H323TLS or Strong security profiles. Also, Communication Manager and the endpoint negotiate by using the H323 TLS profile. H323TLS profile sends H.323 signaling messages through a TLS-encrypted channel.
Allow SIP URI Conversion
Administers whether or not a SIP URI should be permitted to change. Degrading the URI from sips//: to sip//: might result in a less secure call. This is required when SIP SRTP endpoints are allowed to make and receive calls from endpoints that do not support SRTP.
Valid Entry y
Usage Allows conversion of SIP URIs. Default is y.
Table continues...
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Valid Entry n
IP network region: page 3
Usage No URI conversion. Calls from SIP endpoints that support SRTP made to other SIP endpoints that do not support SRTP will fail. However, if you enter y for the Enforce SIPS URI for SRTP field on the signaling group screen, URI conversion takes place independent of the value set for the Allow SIP URI conversion field on the IP Network Region screen.
TCP SIGNALING LINK ESTABLISHMENT FOR AVAYA H.323 ENDPOINTS
Near End Establishes TCP Signaling Socket
Indicates whether Communication Manager (the near end) can establish the TCP socket for H.323 IP endpoints in this network region.
Valid Entry y
n
Usage
Communication Manager determines when to establish the TCP socket with the IP endpoints, assuming the endpoints support this capability. This is the default.
The IP endpoints always attempt to set up the TCP socket immediately after registration. This field should be disabled only in network regions where a nonstandard H.323 proxy device or a non-supported network address translation (NAT) device would prevent the server from establishing TCP sockets with H.323 IP endpoints.
Near End TCP Port Min
Valid Entry 1024 to 65531
Usage
The minimum port value used by the Control Lan (C-LAN) circuit pack or processor Ethernet when establishing the TCP signaling socket to the H.323 IP endpoint. The range of port number must be at least 5 (Max-Min+1). Default is 61440.
Related links Near End TCP Port Max on page 469
Near End TCP Port Max
Valid Entry 1028 to 65535
Usage
The maximum port value to be used by the Control Lan (C-LAN) circuit pack or processor Ethernet when establishing the TCP signaling socket to the H.323 IP endpoint. The range of port number must be at least 5 (Max-Min+1). Default is 61444.
Related links Near End TCP Port Min on page 469
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IP network region
IP network region: page 4
Each subsequent page shows the inter-region connectivity for 15 region pairs.
AGL
The maximum number of destination region IP interfaces included in alternate gatekeeper lists (AGL).
Valid Entry 0 to 16 all
blank
Usage
Communication Manager uses the numeric value of gatekeeper addresses.
Communication Manager includes all possible gatekeeper addresses in the endpoint's own network region and in any regions to which the endpoint's region is directly connected.
The administration field is ignored.
codec-set
Valid Entry 1 to 7 pstn blank
Usage
The codec set used between the two regions. This field cannot be blank if this route through two regions is being used by some non-adjacent pair of regions. If the two regions are disconnected at all, this field should be blank.
direct-WAN
Indicates whether the two regions (source and destination) are directly connected by a WAN link. The default value is enabled if a codec-set is administered.
Related links codec-set on page 470
dst rgn
Valid Entry 1 to 250
Usage The destination region for this inter-network connection.
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IP network region: page 4
Dyn CAC
Use this field to configure a gateway for Call Admission Control (CAC). The system displays this field only if the WAN-BW-limits (Units) field is set to Dynamic.
Valid Entry 1 to 250 blank
Usage The gateway that reports the bandwidth-limit for this link. Default is blank.
Note:
If you set the BW Management Option field to shared-SM, you cannot view this field. Related links
WAN-BW-limits (Units) on page 474
IGAR
Allows pair-wise configuration of Inter-Gateway Alternate Routing (IGAR) between network regions.
Valid Entry y
n
f
Usage
Enables IGAR capability between this network region pair. Default for a pstn codec set.
Disable IGAR capability between this network region pair. Default, except for a pstn codec set.
Forced. Moves all traffic onto the PSTN. This option can be used during initial installation to verify the alternative PSTN facility selected for a network region pair. This option can also be used to temporarily move traffic off of the IP WAN if an edge router is having problems or an edge router needs to be replaced between a network region pair.
Intervening-regions
Allows entry of intervening region numbers between the two indirectly-connected regions.
Valid Entry 1 to 250
Usage Up to four intervening region numbers between the two indirectly-connected regions.
Note:
Indirect region paths cannot be entered until all direct region paths have been entered. In addition, the order of the path through the regions must be specified starting from the source region to the destination region.
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IP network region
Mtce
Valid Entry t m
d
Usage
This is a test-only option. Inter-region connectivity testing is performed for the network region pair by using a simple PING sent between entities in each network region. If a test fails, only an error is added to the system error log. IP media connections between the region pair are never blocked. The testing is done at the rate of not more than once per 5 minutes.
This is a measurement based option. Inter-region connectivity testing is performed by a continuous set of PINGs sent between entities in each network region. The Ping Test Interval (sec) and Number of Pings Per Measurement Interval fields control the rate of testing. The Roundtrip Propagation Delay (ms) and Packet Loss (%) thresholds control success or failure. If the, test measurements exceed the administered thresholds; future IP media connections between the network region pair will be blocked.
No testing is performed for the network region pair.
Related links Ping Test Interval (sec) on page 478 Number of Pings Per Measurement Interval on page 478 Roundtrip Propagation Delay (ms) on page 479 Packet Loss (%) on page 478
src rgn
Valid Entry 1 to 250
Usage The source region for this inter-network connection.
Sync
The system displays Sync when the Synchronization over IP field is enabled.
Valid Entry y
n
Usage
Timing IGC streams are allowed between the region pair that is being administered. The default value is y.
Do not allow timing IGC streams between the region pair that is being administered.
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IP network region: page 4
Video (Norm)
Use this field to specify the amount of bandwidth to allocate for the normal video pool to each IP network region. You can enter:
· 0 to 9999 for Kbits. · 0 to 65 for Mbits. · blank for NoLimit. Note: If you set the BW Management Option field to shared-SM, you cannot view this field.
Video (Prio)
Use this field to specify the amount of bandwidth to allocate for the priority video pool to each IP network region. You can enter:
· 0 to 9999 for Kbits. · 0 to 65 for Mbits. · blank for NoLimit. Note: If you set the BW Management Option field to shared-SM, you cannot view this field.
Video (Shr)
Use this field to specify whether the normal video pool can be shared for each link between IP network regions.
Note: If you set the BW Management Option field to shared-SM, you cannot view this field.
WAN-BW limits (Total)
Use this field to specify the bandwidth limit for direct WAN links. You can enter: · 0 to 9999 for Kbits · 0 to 65 for Mbits · blank for NoLimit
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IP network region
Note: If you set the BW Management Option field to shared-SM, you cannot view this field.
WAN-BW-limits (Units)
Use this field to specify the unit of measurement corresponding to the value entered for bandwidth limitation.
· Calls · Dynamic · Kbits · Mbits · NoLimit Note: If you set the BW Management Option field to shared-SM, you cannot view this field.
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Chapter 70: IP Node Names
Administers node names and IP addresses for the switch and the terminal server media processors administered on the IP Interfaces screen.
Note: The system displays the Processor Ethernet interface node name (procr or procr6) on the IP Node Names screen. The PE interface node name cannot be added to the IP Node Names screen. The line containing the keywords procr or procr6 displays the IP address. Example command: change node-names ip
Name
The name of the adjunct, server, or switch node used as a label for the associated IP address. The node names must be unique for each server or switch. Uses up to 15 alpha-numeric characters.
IP Address
A unique IP address assigned to each port on any IP device that is used for a connection. Also supports IPv6 address format.
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Chapter 71: IP options system parameters
Example command: change system-parameters ip-options
SYSTEM PARAMETERS IP-OPTIONS: page 1
AUTOMATIC TRACE ROUTE ON
Link Failure
Enables or disables the automatic trace route command. If enabled, to diagnose network problems, especially to determine where a network outage exists, Communication Manager initiates an automatic trace-route command when the connectivity between a server and its port networks, gateways, or IP trunks is lost.
Note:
If disabled, any automatic trace-route currently in progress finishes, and no subsequent traceroute commands are started or logged. In other words, the link failure buffer is cleared.
H.248 MEDIA GATEWAY
Link Loss Delay Timeout (minutes)
Valid Entry 1 to 30
Usage
The number of minutes to delay the reaction of the call controller to a link bounce. Assists with the H.248 link bounce recovery mechanism of the Avaya G700 branch gateway. Specifically, prevents the call controller from removing all boards and ports prematurely in response to a link bounce. Default is 5.
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SYSTEM PARAMETERS IP-OPTIONS: page 1
H.323 IP ENDPOINT
Link Loss Delay Timer (minutes)
Valid Entry 1 to 60
Usage
The number of minutes to delay the reaction of the call controller to a link bounce. Specifies how long the Communication Manager server preserves registration and any stable calls that might exist on the endpoint after it has lost the call signaling channel to the endpoint. If the endpoint does not re-establish connection within this period, the system cancels the registration and any calls of the endpoint. This timer does not apply to soft IP endpoints operating in telecommuter mode. Default is 5.
Periodic Registration Timer (min)
This timer is started when an IP telephone registration is taken over by another IP endpoint. When the timer expires, the telephone tries to reregister with the server. Default timer value is dependent on the number of unsuccessful periodic registration attempts. Sample field values apply unless the endpoint is interrupted, such as by power loss, or the user takes manual action to override this automatic process:
· 20 means once every 20 minutes for two hours, then once an hour for 24 hours, then once every 24 hours continually.
· 60 means once an hour for two hours, then once an hour for 24 hours, then once every 24 hours continually.
Valid Entry 1 to 60
Usage
The number of minutes before an IP telephone registration is taken over by another IP endpoint attempts to re-register with the server. Default is 60.
Primary Search Time (seconds)
Valid Entry 15 to 3600
Usage
The maximum number of seconds the IP endpoint attempts to register with its current Communication Manager server while the telephone is disconnected before going to a Survivable Remote Server. With this timer, the customer can specify the maximum time that an IP endpoint spends on trying to connect to the C-LANS before going to a Survivable Remote Server.
When the IP telephone's receiver is lifted, the endpoint continues trying to reestablish connection with the current server until the call ends.
Default is 75.
Short/Prefixed Registration Allowed
The system displays Short/Prefixed Registration Allowed if the IP Stations field on the System Parameters Customer Options screen is set to y.
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IP options system parameters
Valid Entry y
n
Usage
Call Processing allows an IP endpoint to register using a short extension, for the extensions that have Short/Prefixed Registration Allowed field set to default on the Station screen. The default value is y.
Call Processing does not allow an IP endpoint to register using a short extension, for the extensions that have Short/Prefixed Registration Allowed field set to default on the Station screen.
Related links Short/Prefixed Registration Allowed on page 753
IP MEDIA PACKET PERFORMANCE THRESHOLDS
Enable Voice/Network Stats
Enables or disables the recording of voice and network statistics at a system level for all TN2302/ TN2602 media processor boards in the network. The default value is disabled.
Number of Pings Per Measurement Interval
Valid Entry 10 to 100
Usage
the number of test pings that comprise a measurement from which the performance values (delay and loss) are calculated. Default is 10.
Packet Loss (%)
Specifies thresholds to be applied to packet loss rates (as measured by ping) for determining activation or deactivation of signaling group bypass.
High:
Valid Entry 0 to 100
Usage The high value cannot be less than the minimum value. Default is 40.
Low:
Valid Entry 0 to 100
Usage The low value cannot be more than the maximum value. Default is 15.
Ping Test Interval (sec)
Valid Entry 10 to 999
Usage
The time between performance test pings for each testable signaling group. Default is 20.
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SYSTEM PARAMETERS IP-OPTIONS: page 1
Roundtrip Propagation Delay (ms)
Specifies thresholds to be applied to roundtrip packet propagation delays as measured by ping, for use in activating or clearing signaling group bypass.
High:
Valid Entry 10 to 9999
Usage The high value cannot be less than the minimum value. Default is 800.
Low:
Valid Entry 10 to 9999
Usage The low value cannot be more than the maximum value. Default is 400.
MEDIA GATEWAY ANNOUNCEMENT SERVER PARAMETERS
Announcement Server IP Address
The IP address of the Announcement Server that is a unique IP address assigned to each port on any IP device that is used for a connection.
Announcement Storage Path Name
The directory path name on the Announcement Server where the announcements are stored. Accepts up to 40 characters.
Login
The login used by the gateway to access the announcement server. Accepts up to 10 characters.
Password
The password used by the gateway to access the announcement server. Accepts up to 10 characters.
RTCP MONITOR SERVER
The RTCP monitor is a separate computer that receives RTCP packets from many devices. Communication Manager pushes these values to IP telephones, IP softphones and VoIP media modules, such that they know where to send the data.
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IP options system parameters
Default RTCP Report Period (secs)
Valid Entry 5 to 99
Usage
The number of seconds IP telephones, IP softphones, and VoIP media modules send RTCP packets to the RTCP server.
Default Server IP Address
The default IP address of the RTCP server used for each administered region. A unique IP address is assigned to each port on any IP device that is used for a connection.
Related links Server IPV4 Address on page 466
Default Server Port
Valid Entry 1 to 65535
Usage The default TCP/IP port of the RTCP server. Default is 5005.
Link Failure
Enables or disables the automatic trace route command. If enabled, to diagnose network problems, especially to determine where a network outage exists, Communication Manager initiates an automatic trace-route command when the connectivity between a server and its port networks, gateways, or IP trunks is lost.
Note:
If disabled, any automatic trace-route currently in progress finishes, and no subsequent traceroute commands are started or logged. In other words, the link failure buffer is cleared.
SYSTEM PARAMETERS IP-OPTIONS: page 2
Prefer use of G.711 by Music Sources
Enables or disables using G.711 for intra-switch Music-On-Hold. The default value is n.
Force Phones and Gateways to Active LSPs
Enables or disables forcing telephones and gateways to active LSPs. The default is disabled.
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SYSTEM PARAMETERS IP-OPTIONS: page 2
HYPERACTIVE MEDIA GATEWAY REGISTRATIONS
Enable Detection and Alarms
Enables or disables the hyperactive gateway registration feature. Default is disabled.
Number of Registrations within the Window
Valid Entry 1 to 19
Usage
The number of registrations that occur within the hyperactivity window for generating a Gateway alarm. Default is 3. Available only if detection and alarming is enabled.
Related links Enable Detection and Alarms on page 481
Parameters for Media Gateway Alarms: Hyperactive Registration Window (minutes)
Available only if detection and alarming is enabled.
Valid Entry 1 to 15
Usage
The time in minutes for checking hyperactive gateway registrations. Default is 4 minutes.
Related links Enable Detection and Alarms on page 481
Parameters for Network Region Registration (NR-REG) Alarms: % of Gateways in Network Region with Hyperactive Registration Alarms
Available only if detection and alarming is enabled.
Valid Entry 1 to 99
Usage
The percent of Gateways within an ip-network region that should be alarmed before an IP-Registration alarm is generated. Default is 80%.
Related links Enable Detection and Alarms on page 481
IP DTMF TRANSMISSION MODE
Inter-System IP DTMF Transmission Mode
Specifies the touchtone signals that are used for dual-tone multifrequency (DTMF) telephone signaling.
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IP options system parameters
Valid Entry in-band in-band-g711 out-of-band
rtp-payload
Usage
All G711 and G729 calls pass DTMF in-band. DTMF digits encoded within existing RTP media stream for G.711/G.729 calls. G.723 is sent out-of-band. It also supports the SIP trunks.
Only G711 calls pass DTMF in-band. The system displays the in-band-g711 option if the Group Type field is set to h.323.
All IP calls pass DTMF out-of-band. For IP trunks, the digits are done with either Keypad IEs or H245 indications. This is the default for newly added H.323 signaling groups.
The out-of-band option enables out of band signaling for the SIP Signaling group. When you select the out-of-band option, Communication Manager sends all outgoing DTMF messages as SIP INFO messages over a SIP signaling group.
This option is interoperable with H.323 networks. You can connect an Avaya nonSIP endpoint or trunk to a voice mail system on the H.323 network that is integrated with Session Manager and Communication Manager through SIP. Examples of Avaya non-SIP endpoints or trunks are H.323, Analog, DCP, and Digital trunks.
This is the method specified by RFC 2833. This is the default value for newly added SIP signaling groups. Support for SIP trunks requires the default entry of rtppayload.
Intra-System IP DTMF Transmission Mode
The IP transmission mode.
Valid Entry in-band
rtp-payload
Usage
DTMF digits encoded within existing RTP media stream for G.711/G.729 calls. G.723 is sent out-of-band.
Support for SIP trunks requires the entry of rtp-payload.
SYSTEM PARAMETERS IP-OPTIONS: page 3
ALTERNATIVE NETWORK ADDRESS TYPES
To control the call processing behavior to send Alternative Network Address Types (ANAT) offer system wide.
ANAT Enabled
Use this field to control the call processing behavior to send Alternative Network Address Types (ANAT) offer system wide.
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Valid entry y
n
SYSTEM PARAMETERS IP-OPTIONS: page 3
Usage Communication Manager sends ANAT offer to the SIP elements system wide. Communication Manager does not send ANAT offer to the SIP elements system wide.
SERVICES DIALPAD PARAMETERS
To disable the services dialpad parameters on the telephones, set the Download Flag field to y and leave the Password field blank.
Download Flag
If you set this field to y, Communication Manager downloads the services dialpad parameters and the associated IP addresses to the telephones. The default value is n.
Password
Use this field to enter the password that a technician uses to administer craft procedures. During the installation or after the successful installation of an IP telephone, the technician performs craft procedures to customize the telephone installation for specific operating environment. You can provide a password of length up to seven digits or leave the field blank. The default value is 27238.
SNMP STATION PARAMETERS
To disable the SNMP parameters on the telephones, set the Download Flag field to y and leave the Community String field blank.
Community String
Use this field to enter a string that an IP telephone uses to determine whether the telephone must receive SNMP messages. You can enter up to 32 characters.
The telephone does not respond to the incoming SNMP message if you:
· leave the Community String field blank and the source address of the SNMP message does not match with an address in the SNMP Source Address list
· leave the Community String field blank and the source address of the SNMP message matches with an address in the SNMP Source Address list
· set the Community String field to a value and the source address of the SNMP message does not match with an address in the SNMP Source Address list
·
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IP options system parameters
The telephone responds to the incoming SNMP message only if you set the Community String field to a value and the source address of the SNMP message matches with an address in the SNMP Source Address list.
Download Flag
If you set this field to y, Communication Manager downloads the SNMP parameters and the associated IP addresses to the telephones. The default value is n.
SOURCE ADDRESSES
Use this field to enter node names, mapping to proper IP address, that an IP telephone uses to validate the source address of an SNMP message. The telephone does not respond to the incoming SNMP message if you:
· leave the Community String field blank and the source address of the SNMP message does not match with an address in the SNMP Source Address list
· leave the Community String field blank and the source address of the SNMP message matches with an address in the SNMP Source Address list
· set the Community String field to a value and the source address of the SNMP message does not match with an address in the SNMP Source Address list
· The telephone responds to the incoming SNMP message only if you set the Community String field to a value and the source address of the SNMP message matches with an address in the SNMP Source Address list. You can enter up to six node names. Use the IP Node Names screen to administer node names. Related links IP Address on page 475
SYSTEM PARAMETERS IP-OPTIONS: page 4
Dest # 1, 2, or 3 IP address
The valid destination IPv4 address. The default destination address is 0.0.0.0.
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SYSTEM PARAMETERS IP-OPTIONS: page 4
Local Facility
Valid Entry local0 to local7
Usage
Displays the help message upon acceptable values for local use. The default value is local4.
Port #
Valid Entry 1 to 65535
Usage
The valid port number associated with the destination IPv4 address. The default port number is 514.
Related links Dest # 1, 2, or 3 IP address on page 484
MUSIC/ANNOUNCEMENTS IP-CODEC PREFERENCES
In general, when operating across the WAN with limited bandwidth facilities, the ip-codec-set is configured only with the compressed voice codecs. Sometimes it is necessary to carry the voice calls with the normal configured compressed codecs, and the music or announcement sources are received with non-compressed G.711 codecs.
If you administer the following fields, when possible, the system overrides the ip-codec-set preference which can be configured to prefer a compressed codec, with non-compressed G.711. If the device receives the music or announcement source which has signaled support for G.711, the system attempts to use G.711.
Prefer use of G.711 by Announcement Sources
Overrides the ip-codec-set preference and establishes inter-PN or inter-gateway connections transmitting announcements with G.711. The default value is n.
Prefer use of G.711 by IP Endpoints Listening to Announcements
Overrides the ip-codec-set preference and reconfigures the IP endpoints listening to an announcement source with G.711. The default value is n.
Prefer use of G.711 by IP Endpoints Listening to Music
Overrides the ip-codec-set preference and reconfigures the IP endpoints listening to a music source with G.711. The default value is n.
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IP options system parameters
Prefer use of G.711 by Music Sources
Overrides the ip-codec-set preference and establishes inter-PN or inter-gateway connections transmitting music with G.711. The default value is n.
SYSTEM PARAMETERS IP-OPTIONS: page 5
BW Management Option
Use this field to select the bandwidth management mode.
Valid entries local-CM
shared-SM
Usage
With this mode, Communication Manager acts as stand-alone bandwidth management entity. Bandwidth limits for the bandwidth used by Communication Manager are set on Communication Manager.
The default value is local-CM. With this mode, Session Manager acts as the central authority for bandwidth management and Communication Manager obtains bandwidth for voice and multimedia IP connections from Session Manager. Bandwidth limits for the bandwidth collectively used by Communication Manager and other users are set using System Manager. For more information, see Administering Avaya Aura® Session Manager.
Signaling-group of Primary SM BW Mgr
Use this field to specify the assigned Session Manager SIP signaling group number. You can enter a value from 1 to 999. Do not leave the field blank. The system displays the field only when you set the BW Management Option field to sharedSM.
Signaling-group of Secondary SM BW Mgr
Use this field to specify the assigned Session Manager SIP signaling group number. You can enter a value from 1 to 999. You can leave the field blank. This optional secondary Session Manager SIP signaling group is only used when the primary Session Manager SIP signaling group is unavailable. The secondary Session Manager SIP signaling group cannot be the same number as the primary Session Manager SIP signaling group.
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SYSTEM PARAMETERS IP-OPTIONS: page 5
The system displays the field only when you set the BW Management Option field to sharedSM.
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Chapter 72: IP Routing
There is one-to-one mapping between the Network Bits and the Subnet Mask fields; entering a value in one field uniquely determines the other field. A list of Subnet Mask addresses and their corresponding Network Bits are shown in Table below.
Network Bits 0 1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24
Subnet Mask 0.0.0.0 128.0.0.0 192.0.0.0 224.0.0.0 240.0.0.0 248.0.0.0 252.0.0.0 254.0.0.0 255.0.0.0 255.128.0.0 255.192.0.0 255.224.0.0 255.240.0.0 255.248.0.0 255.252.0.0 255.254.0.0 255.255.0.0 255.255.128.0 255.255.192.0 255.255.224.0 255.255.240.0 255.255.248.0 255.255.252.0 255.255.254.0 255.255.255.0
Number of Hosts 4,294,967,294 2,147,483,646 1,073,741,822 536,870,910 268,435,454 134,217,726 67,108,862 33,554,430 16,777,214 8,388,606 4,194,302 2,097,150 1,048,574 524,286 262,142 131,070 65,534 32,766 16,382 8,190 4,094 2,046 1,022 510 254
Network Type / 0 / 1 / 2 / 3 / 4 / 5 / 6 / 7 / 8 / 9 / 10 / 11 / 12 / 13 / 14 / 15 / 16 / 17 / 18 / 19 / 20 / 21 / 22 / 23 / 24
Table continues...
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Network Bits 25 26 27 28 29 30 31
Subnet Mask 255.255.255.128 255.255.255.192 255.255.255.224 255.255.255.240 255.255.255.248 255.255.255.252 255.255.255.254 255.255.255.255
Example command: add ip-route next
Number of Hosts 126 62 30 14 6 2 1 0
Board
Network Type / 25 / 26 / 27 / 28 / 29 / 30 /31 /32
Board
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks) 0131 1 to 250 V1 to V9
Usage First and second characters are the cabinet number Third character is the carrier Fourth and fifth character are the slot number Six and seventh characters are the circuit number
Gateway Module
Destination Node
The name of the final destination node of the IP route for this connection. Related links
IP Node Names on page 475
Gateway
The node name of the first intermediate node consisting of a port on the CLAN circuit pack or a Destination Node on another IP route. If there are one or more intermediate nodes, the first
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IP Routing
intermediate node is the Gateway. If there are no intermediate nodes between the local and remote CLAN ports for this connection, the Gateway is the local CLAN port. Related links IP Node Names on page 475
Metric
Valid Entry 0 1
Usage A server that has only one CLAN circuit pack installed. A server that has more than one CLAN circuit pack installed.
Network Bits
A 32-bit binary number that divides the network ID and the host ID in an IP address.
Valid Entry 0 to 32
Usage The number of Network Bits associated with this IP route.
Route Number
Valid entry digits
Usage
The IP route number. The upper limit can change depending on the system configuration. For more information, see Avaya Aura® Communication Manager System Capacities Table.
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Chapter 73: IP Server Interface (IPSI) Administration
Adds a TN2312 IPSI (IP Server Interface) circuit pack. Uses the IP Server Interface (IPSI) to control port networks and provide tone, clock, and call classification services. The IPSI board connects to the control network by way of Ethernet. In Communication Manager Release 5.2, the IP server interface administration for the TN2312 IPSI or the TN8412 SIPI provides support for Communication Manager--based SAT administration of IPSI Quality of Service (QoS) and Ethernet interface settings parameters. All further references to IPSI also apply to the TN8412 SIPI.
Note: Initial IPSI settings must be done using the IPSI CLI interface. Example command: change ipserver-interface n, where n is the assigned IPSI board location.
IP Server Interface (IPSI) Administration: page 1
Administer secondary ip server interface board
If enabled, assigns a secondary IPSI board.
Enable QoS
If enabled, turns on quality of service (QoS) from the server to the IPSI link. If you enable QoS for the control network, also enable it from the Web interface.
Encryption
Enables socket encryption for the server and IPSI link.
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IP Server Interface (IPSI) Administration
Ignore Connectivity in Server Arbitration
If enabled, does not test the integrity of the IPSI when checking the health of the server pair.
IP Control
Administers IP control of port networks.
Valid Entry y
n
Usage
All port networks have an IPSI that provides control. A DS1 Converter (DS1C) circuit pack cannot be added to a port network when this value is y.
This IPSI is used only for Tone Clock/Tone Detector functions. Use when the port network contains a DS1 Converter (DS1C) circuit pack.
PRIMARY IPSI
DHCP
Displays whether IPSI is currently set up for DHCP addressing, or static addressing.
· If DHCP is disabled for the System Management Interface, this field is set to disabled (readonly).
· If DHCP is disabled for the IP Server Interface, the following event occurs:
- If IPSI is in-service, it disallows the service and displays the ipserver must be busied out message.
- If IPSI is busied out, the static equivalent to the DHCP address automatically populates the Host. The Subnet Mask and Gateway values must be populated manually. The prepopulated values can be optionally overwritten.
· If DHCP is enabled for IP Server Interface, the following event occurs:
- If IPSI is in-service, it disallows the service and displays the ipserver must be busied out message.
- If the If the IPSI is busied out, it accepts the changes and re-populates this field.
Gateway
The valid gateway IPv4 address.
· If DHCP is enabled for the IP Server Interface, the value of this field is read-only. View the value of this field based on the access.
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IP Server Interface (IPSI) Administration: page 1
· If DHCP is disabled for the IP Server Interface and this field is changed, after screen validation, the system checks if IPSI is busied out. If not, the IPSI does not accept the change and displays the ipserver must be busiedout message.
Host
The name of the DHCP client identifier. If DHCP is enabled for the System Management Interface, the Host value is displayed for the IP Server Interface. If DHCP is disabled, the Host value is not displayed.
IP Address
The valid IPv4 IP address.
· If DHCP is enabled for the IP Server Interface, the value of this field is read-only. View the value of this field based on the access.
· If DHCP is disabled for the IP Server Interface and this field is changed, after screen validation, the system checks if IPSI is busied out. If not, the IPSI does not accept the change and displays the ipserver must be busiedout message.
Location
The primary IPSI board location.
Valid Entry 01 to 64 A to E 01 to 20 01 to 250 V1 to V9
Usage First and second numbers are the cabinet number Third character is the carrier Fourth and fifth characters are the slot number Gateway Module
Subnet Mask
Valid Entry /xx
Usage
Represented as subnet bits.
· If DHCP is enabled for the IP Server Interface, the value of this field is read-only. View the value of this field based on the access.
· If DHCP is disabled for the IP Server Interface and this field is changed, after screen validation, the system checks if IPSI is busied out. If not, the IPSI does not accept the change and displays the ipserver must be busiedout message.
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IP Server Interface (IPSI) Administration
QoS AND ETHERNET SETTINGS
802.1p
Valid Entry 0 to 7
Usage
Displays the 802.1p value. This value can be changed only if the System Level Parameter Values is disabled for the IP Server interface. This value takes effect when the IPSI is busied out or released. Accepts only whole numbers.
The default value is 6.
Related links Use System Level Parameter Values on page 494
Auto
If the IPSI is busied out, you can change this value from disabled to enabled and enabled to disabled.
DiffServ
Valid Entry 0 to 63
Usage
Displays the DiffServ code point (DSCP). This value can be changed only if the System Level Parameters Values is disabled for the IP Server interface. This value takes effect when the IPSI is busied out or released. Default is 46. Accepts only whole numbers.
Related links Use System Level Parameter Values on page 494
Duplex
Valid Entry Half Full
Usage
Displays the duplex settings for this IP board. This value can be changed only if the IPSI is busied out. The default is Full.
Speed
Valid Entry 10Mbps 100Mbps
Usage
Displays the speed of the Ethernet connection. This value can be changed only if the IPSI is busied out. The default value is 100 Mbps.
Use System Level Parameter Values
Enables or disables system level parameter values.
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IP Server Interface (IPSI) Administration: page 1
SECONDARY IPSI
DHCP
Displays whether IPSI is currently set up for DHCP addressing, or static addressing. · If DHCP is disabled for the System Management Interface, this field is set to disabled (readonly). · If DHCP is disabled for the IP Server Interface, the following event occurs: - If IPSI is in-service, it disallows the service and displays the ipserver must be busied out message. - If IPSI is busied out, the static equivalent to the DHCP address automatically populates the Host. The Subnet Mask and Gateway values must be populated manually. The prepopulated values can be optionally overwritten. · If DHCP is enabled for IP Server Interface, the following event occurs: - If IPSI is in-service, it disallows the service and displays the ipserver must be busied out message. - If the If the IPSI is busied out, it accepts the changes and re-populates this field.
Host
The name of the DHCP client identifier. If DHCP is enabled for the System Management Interface, the Host value is displayed for the IP Server Interface. If DHCP is disabled, the Host value is not displayed.
Gateway
The valid gateway IPv4 address. · If DHCP is enabled for the IP Server Interface, the value of this field is read-only. View the value of this field based on the access. · If DHCP is disabled for the IP Server Interface and this field is changed, after screen validation, the system checks if IPSI is busied out. If not, the IPSI does not accept the change and displays the ipserver must be busiedout message.
IP Address
The valid IPv4 IP address. · If DHCP is enabled for the IP Server Interface, the value of this field is read-only. View the value of this field based on the access. · If DHCP is disabled for the IP Server Interface and this field is changed, after screen validation, the system checks if IPSI is busied out. If not, the IPSI does not accept the change and displays the ipserver must be busiedout message.
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IP Server Interface (IPSI) Administration
Location
The secondary IPSI board location.
Valid Entry 01 to 64 A to E 01 to 20 01 to 250 V1 to V9
Usage First and second numbers are the cabinet number Third character is the carrier Fourth and fifth characters are the slot number Gateway Module
Subnet Mask
Valid Entry /xx
Usage
Represented as subnet bits.
· If DHCP is enabled for the IP Server Interface, the value of this field is read-only. View the value of this field based on the access.
· If DHCP is disabled for the IP Server Interface and this field is changed, after screen validation, the system checks if IPSI is busied out. If not, the IPSI does not accept the change and displays the ipserver must be busiedout message.
IP Server Interface (IPSI) Administration: page 2
Dest # 1, 2, or 3 IP Address
The valid destination IPv4 IP address format. Provides support for future IPv6 address format. Default destination address is unspecified.
Enable Syslog
Enabled only for init and inads login access. For duplicated TN2602 boards, this field is displayed for each board.
Valid Entry y
n
Usage
The system checks the firmware support syslog by capabilities exchange information. If it does not get the support, the system displays the unsupported board code or vintage error message and does not enable syslog.
The system does not display the rest of the syslog fields. This is the default.
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IP Server Interface (IPSI) Administration: page 2
Local Facility #
Valid Entry
local0 to local7 (for local use)
Usage Displays the help message on acceptable values. Default value is local4.
Port #
Valid Entry 1 to 65535
Usage
The valid port number associated with the Dest # 1, 2 or 3 IP address value. Default port number is 514.
Related links Dest # 1, 2, or 3 IP Address on page 496
Use System Syslog Values
If enabled, populates the address, port, and facility information as read-only.
DEBUG FILTER VALUES
Object
Valid Entry 0 to 255
Usage The category of a log event. For example, angel and archangel.
Note: Maximum value of 255 triggers a file dump.
Level
Valid Entry 0 to 65535
Usage The level of logging for the given object.
Note: Maximum value of 65535 triggers a file dump.
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Chapter 74: IP Services
Administers the connectivity for various adjuncts. Note: You cannot remove a service from this screen if that service has overrides defined on the Survivable Processor screen.
Example command: change ip-services
IP Services: page 1
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IP Services: page 1
Enabled
Enables or disables this IP service. Available only with an IP Service Type of AESVCS or SAT. Related links
Service Type on page 500
Local Node
Valid Entry IP node name
procr
Usage
A previously-administered node name for the link administered for services over the Control Lan (C-LAN) circuit pack.
The Communication Manager's Processor Ethernet interface for adjunct connectivity.
Related links Name on page 475
Local Port
The originating port number.
Valid Entry 5000 to 9999
0
Usage 5111 to 5117 for SAT applications. 8765 for AE Services. For client applications, defaults to 0.
Remote Node
The server or switch at the far end of the link for SAT.
Valid Entry IP node name any
Usage A previously-administered node name used to provide added security for SAT. Any available node.
Remote Port
The port number of the destination.
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IP Services
Valid Entry 5000 to 64500
0
Usage
If this service is a client application, such as CDR or PMS. This value must match the port administered on the adjunct, Personal Computer, or terminal server that is at the remote end of this connection.
Default for System Management applications.
Service Type
The service provided.
Valid Entry AESVCS CDR1, CDR2
PMS PMS_JOURNAL PMS_LOG SAT SYS_PRINT
Usage
AE Services.
Either the primary or secondary CDR device connects over a TCP/IP link or TLS link.
If you enter y for the TLS Encryption, then the primary or secondary CDR device connects over a TLS link. Else, the primary or secondary CDR device connects over a TCP/IP link.
Property Management System.
The PMS journal printer connects over a TCP/IP link.
The PMS log printer connects over a TCP/IP link.
System administration terminal.
The system printer connects over a TCP/IP link.
TLS Encryption
Allows you to configure TLS encryption for CDR1 and CDR2 service types.
Valid Entry Y N
Usage
Enables TLS encryption for CDR1 and CDR2 service types.
Disables TLS encryption. Entering n supports backward compatibility where TCP only is used. This is the default value.
IP Services: page 3
Enables reliable protocol for TCP/IP links, and establishes other session-layer parameters. Available only if CDR1, CDR2, PMS_JOURNAL, or PMS_LOG is administered for the Service Type.
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IP Services: page 3
Connectivity Timer
Valid Entry 1 to 255
Usage
The amount of time in seconds that the link can be idle before Communication Manager sends a connectivity message to ensure the link is still up.
Packet Resp Timer
Valid Entry 1 to 255
Usage
The number of seconds to wait from the time a packet is sent until a response (acknowledgement) is received from the far-end, before trying to resend the packet.
Reliable Protocol
Enables or disables reliable protocol over this link. If enabled, uses reliable protocol if the adjunct on the far end of the link supports it.
Service Type
The service type for establishing parameters.
Valid Entry CDR1, CDR2 PMS_JOURNAL PMS_LOG
Usage Connects either the primary or secondary CDR device over a TCP/ IP link. Connects the PMS journal printer over a TCP/IP link. Connects the PMS log printer over a TCP/IP link.
Session Connect Message Cntr
Valid Entry 1 to 5
Usage
The number of times Communication Manager tries to establish a connection with the far-end adjunct.
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IP Services
SPDU Cntr
Valid Entry 1 to 5
Usage
The number of times Communication Manager transmits a unit of protocol data before generating an error.
IP Services: page 4
Creates symbolic name and password pairs for all AE Services servers that are allowed to connect to Communication Manager. Available only if the Service Type is AESVCS.
Enabled
Enables or disables the AE Services server.
Password
A password for future access. Accepts 12 to 16 alphanumeric characters and must contain at least one alpha character and one numeric character.
AE Services Server
A valid AE Services Server name. The name must match the AE Services server machine name. Each name must be unique.
Server ID
Valid Entry 1 to 16
Usage The number assigned to this server.
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IP Services: page 4
Status
Valid Entry idle in-use blank
Usage The AE Services server is connected to Communication Manager. The AE Services server is disconnected to Communication Manager. No AE Server is administered.
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Chapter 75: ISDN Trunk Group
Assigns an Integrated Services Digital Network (ISDN) trunk group that supports the ISDN and Callby-Call Service Selection service selection features. The trunk group provides end-to-end digital connectivity and supports a wide range of services including voice and non-voice services, to which users have access by a limited set of CCITT-defined, standard multipurpose interfaces. The ISDN trunk group can contain ISDN-PRI or ISDN-BRI interfaces. However, it is not possible to use the two types of interfaces in the same trunk groups. The type of interface is chosen when the trunk members are assigned to the trunk group. When ISDN-PRI interfaces are used on ISDN trunk groups, they can also be used to support the Wideband Switching feature. This is intended to work with the H0 (384 Kbps), H11 (1536 Kbps), H12 (1920 Kbps), and NXDS0 (128 to 1984 Kbps) data services, and to support high-speed video conferencing and data applications. Example command: add trunk-group n, where n is the trunk group number.
ISDN Trunk Group: page 1
Auth Code
The Auth Code field is available only for incoming or two-way trunk groups if you enable the Authorization Codes feature. If you enable the Auth Code field, Communication Manager performs an auth code check for the incoming trunk call that is routed over another trunk.
The following table provides the situations when the caller must enter an auth code.
Calling party Station
Called party
Station Trunk
Facility Restriction Level (FRL) check?
No
Yes, if Route Pattern accesses the trunk
Incoming trunk Station
No
Auth Code, required if
No the FRL of the calling station is less than the FRL of the outgoing route pattern preference the Auth Code field is enabled on the incoming trunk group
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Calling party
Called party Trunk
Facility Restriction Level (FRL) check?
Yes, if Route Pattern accesses the trunk
Auth Code, required if
the FRL of the incoming trunk is less than the FRL of the outgoing route pattern preference or the Auth Code field is enabled on the incoming trunk group
Note: The Auth Code field is unavailable if:
· the Group Type field is tandem · the Group Type field is ISDN and the Service Type field is tandem. In these situations, permissions of the caller are transmitted using Traveling Class Mark. Related links Direction on page 507 Authorization Codes on page 831
Busy Threshold
Valid Entry 1 to 255
Usage
The number of trunks that must be busy to alert attendants to control access to outgoing and two-way trunk groups during periods of high use. When the threshold is reached and the warning lamp for that trunk group lights, the attendant can activate trunk group control: internal callers who dial out using a trunk access code are connected to the attendant. Calls handled by AAR and ARS route patterns go out normally.
Carrier Medium
The type of transport medium interface used for the ISDN trunk group.
Valid Entry ATM
H.323 PRI/BRI
Usage
The trunk is implemented via the ATM Interface circuit pack. Available only if ATM trunking is enabled for the system.
The trunk is implemented as an H.323 trunk group.
The trunk is implemented as a standard DS1 or BRI interface.
Related links Asynch. Transfer Mode (ATM) Trunking on page 830 SBS on page 530
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Local Country Code on page 322 International Access Code on page 323 Group Type on page 508 Supplementary Service Protocol on page 521
CDR Reports
Valid Entry y
n r (ring-intvl)
Usage
All outgoing calls on this trunk group generate call detail records. If Record Outgoing Calls Only is disabled for CDR, incoming calls on this trunk group also generate call detail records.
Calls over this trunk group do not generate call detail records.
CDR records are generated for both incoming and outgoing calls. In addition, the following ringing interval CDR records are generated:
· Abandoned calls: The system creates a record with a condition code of "H" indicating the time until the call was abandoned.
· Answered calls: The system creates a record with a condition code of "G" indicating the interval from start of ring to answer.
· Calls to busy stations: The system creates a record with a condition code of " I " indicating a recorded interval of 0.
Related links Record Outgoing Calls Only on page 111
Charge Advice
Determines how to accumulate and access charge information about a call. Requires that CDR Reports be enabled before changing this field from its default of none. Receiving Advice of Charge during the call affects system performance because of the increased ISDN message activity on the signaling channel, which might reduce the maximum call capacity.
Valid Entry none automatic end-on-request
during-on-request
Usage
The system does not to collect Advice of Charge information for this trunk group.
The public network sends Advice of Charge information automatically.
Communication Manager requests charge information with each call, and you receive only the final call charge.
Communication Manager requests charge information with each call, and charges display during and at the end of a call.
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COR
Valid Entry 0 to 995
Usage
The Class of Restriction (COR) for the trunk group. Classes of restriction control access to trunk groups, including trunk-to-trunk transfers. Decisions regarding the use of Class of Restriction (COR) and Facility Restriction Levels (FRLs) should be made with an understanding of their implications for allowing or denying calls when AAR/ARS/WCR route patterns are accessed.
Tip:
Remember that FRLs are assigned to classes of restriction. Even if two trunk groups have classes of restriction that allow a connection, different facility restriction levels might prevent operations such as off-net call forwarding or outgoing calls by remote access users.
Dial Access
Controls whether users can route outgoing calls through an outgoing or two-way trunk group by dialing its trunk access code. Allowing dial access does not interfere with the operation of AAR/ ARS.
Security alert:
Calls dialed with a trunk access code over WATS trunks bypass AAR/ARS and are not restricted by facility restriction levels. For security, leave this field disabled unless dial access is needed to test the trunk group.
Valid Entry y n
Usage
Users can gain access to the trunk group by dialing its access code.
Users cannot gain access to the trunk group by dialing its access code. Attendants can still select this trunk group with a Trunk Group Select button.
Direction
The direction of the traffic on this trunk group. Available for all trunk groups except DID and CPE.
Valid Entry incoming outgoing two-way
Usage
Traffic on this trunk group is incoming.
Traffic on this trunk group is outgoing
Traffic on this trunk group is incoming and outgoing. Required for Network Call Redirection.
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ISDN Trunk Group
Related links Disconnect Supervision-Out on page 519 Answer Supervision Timeout on page 517 Receive Answer Supervision on page 649
Far End Test Line No.
The number sent to the far-end's ISDN test line extension. When the test trunk long command is issued, this exact number is sent to the far-end to establish a call that tests the integrity of the trunk member under test. The number does not pass through routing or undergo digit manipulation. The digits entered must be what the far-end expects. Accepts up to 15 digits. Example For an ISDN tandem trunk, the far-end test number should be a seven-digit ETN (Electronic Tandem Network) number.
Group Name
A unique name that provides information about the trunk group. Accepts up to 27 characters. This field should contain names that identify the vendor and function of the trunk group rather than the group type (DID, WATS).
Note: Supported by Unicode language display for the 4610SW, 4620SW, 4621SW, and 4622SW, 1600-Series, 9600-Series IP Deskphones, 9600-Series H.323 Deskphones, and Avaya J100 Series IP Phones. Unicode is also an option for the 2420J telephone when the Display Character Set is katakana. For more information about 2420J, see 2420 Digital Telephone User's Guide. Related links Display Character Set on page 821
Group Number
The trunk group number.
Group Type
The type of trunk group. The fields that are displayed and available might change according to the trunk group type selected.
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Valid Entry Access APLT CAMA CO CPE DID DIOD DMI-BOS FX ISDN
RLT SIP
Tandem
ISDN Trunk Group: page 1
Usage
Used to connect satellite servers to the main switch in Electronic Tandem Networks (ETN). Access trunks do not carry traveling class marks (TCM) and, therefore, allow satellite callers unrestricted access to out-dial trunks on the main server. This entry allows Inband ANI.
Advanced Private Line Termination (APLT) trunks. Used in private networks. This entry allows Inband ANI.
Used to route emergency calls to the local community's Enhanced 911 systems.
Typically used to connect Communication Manager to the local telephone company central office, but can also connect adjuncts such as external paging systems and data modules.
Used to connect adjuncts, such as paging systems and announcement or music sources, to the server running Communication Manager.
Used to direct callers directly to individuals within an organization without going through an attendant or some other central point. This entry allows Inband ANI.
Two-way trunks that are used to transmit dialed digits in both directions. In North America, tie trunks are used for applications that require two-way transmission of dialed digits. This entry allows Inband ANI.
Digital Multiplexed Interface - Bit-Oriented Signaling (DMI-BOS) trunks allow communication with systems using DMI-BOS protocol. This entry also allows Inband ANI.
A local telephone company central office (CO) trunk that connects the server running Communication Manager directly to a CO outside the local exchange area. Used to reduce long-distance charges if the organization averages a high volume of longdistance calls to a specific area code.
Used when digital trunks are needed that can integrate voice, data, and video signals and provide the bandwidth needed for applications such as high-speed data transfer and video conferencing. ISDN trunks can also efficiently combine multiple services on one trunk group.
Also used for Network Call Transfer.
Note:
Available only if ISDN-PRI, ISDN-BRI Trunks, or both have been enabled for the system. For descriptions of the screens and fields that are unique to ISDN trunks, see "ISDN Trunk Group".
Used with Centralized Attendant Service in a private network.
Used to connect a server running Communication Manager to a Session Manager (SM), or to connect two Communication Manager servers.
Note:
The Automatic CallBack, Priority Calling, and Whisper Page features work incorrectly if each of the call's parties is using a SIP endpoint administered on and managed by a different instance of Communication Manager.
Used to connect tandem nodes in a private network. This entry allows Inband ANI.
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ISDN Trunk Group
Valid Entry Tie
WATS
Usage
Used to connect a server running Communication Manager to a local telephone company central office or to another server or switch in a private network. Tie trunks transmit dialed digits with both outgoing and incoming calls. This entry also allows Inband ANI.
Used to reduce long-distance bills when your organization regularly places many calls to a specific geographical area in North America. With outgoing WATS service, calls can be sent to certain areas ("WATS band") for a flat monthly charge. Incoming WATS trunks allow toll-free calling to customers and employees.
Related links ISDN-PRI on page 836 ISDN-BRI Trunks on page 835 SBS on page 530 Local Country Code on page 322 International Access Code on page 323 Supplementary Service Protocol on page 521 Carrier Medium on page 505 Path Replacement on page 536 Call Still Held on page 922 Call Still Held on page 644
Incoming Calling Number - Format
The TON/NPI encoding applied to CPN information modified by the CLI Prefix feature. This encoding does not apply to calls originating locally.
If this field is blank, Communication Manager passes on the encoding received in the incoming setup message. If the incoming setup message excludes CPN information and digits are added, the outgoing message contains these digits. If a numbering format is not administered in this case, the value defaults to pub-unk. If the numbering format is administered as unknown, the trunk group is modified to unk-unk encoding of the TON/NPI. Therefore, this field also must contain a value other than unknown.
The values for this field map to the Type of Numbering (TON) and Numbering Plan Identifier (NPI) values shown below.
Valid Entry blank natl-pub intl-pub locl-pub
Type of Numbering (TON) incoming TON unmodified national(2) international(1) local/subscriber(4)
Numbering Plan Identifier (NPI) incoming NPI unmodified E.164(1) E.164(1) E.164(1)
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Valid Entry pub-unk lev0-pvt lev1-pvt lev2-pvt unk-unk
Type of Numbering (TON) unknown(0) local(4) Regional Level 1(2) Regional Level 2(1) unknown(0)
Numbering Plan Identifier (NPI) E.164(1) Private Numbering Plan - PNP(9) Private Numbering Plan - PNP(9) Private Numbering Plan - PNP(9) unknown(0)
Related links Format on page 927 Format on page 525
Incoming Calling Number Insert
Valid Entry 0 to 9 all blank
Usage
The number of digits inserted in the calling party number for all incoming calls on this trunk group. Accepts up to 15 characters.
Member Assignment Method
Available only if the Carrier Medium is H.323 and SIP.
Valid Entry manual auto
Usage Users manually assign trunk members to a signaling group. This is the default. The system automatically generates members to a specific signaling group.
Related links Carrier Medium on page 505
Number of Members
Available only if the Carrier Medium is H.323 or SIP and the Member Assignment Method is auto.
Valid Entry 0 to 255
Usage
The number of virtual trunk members automatically assigned to the signaling group. Default is 0.
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ISDN Trunk Group
Related links Carrier Medium on page 505 Member Assignment Method on page 511 Signaling Group on page 514
Outgoing Display
With Outgoing Display, telephones can show the name and number of the trunk group used for an outgoing call before the call is connected.
Valid Entry y n
Usage Displays the trunk group name and number. Displays the digits the caller dials.
Queue Length
Available only for outgoing or two-way trunk groups.
Valid Entry 1 to 100
0
Usage
The number of outgoing calls that can wait in queue when all trunks in a trunk group are busy. Calls wait in queue in the order in which they were made. If a queue is administered, a caller hears a confirmation tone when no trunk is available for the outgoing call. The caller can then disconnect and wait; when a trunk becomes available, Communication Manager calls the extension that placed the original call. Communication Manager remembers the number the caller dialed and automatically completes the call.
Callers receive a busy signal when no trunks are available. Use for DCS trunks.
Related links Direction on page 507
Service Type
The service for which this trunk group is dedicated. In addition to the predefined services or features listed as valid entries, any previously administered user-defined Network Facility Facility Type of 0 (feature) or 1 (service) is allowed.
Valid Entry access
Usage A tie trunk giving access to an Electronic Tandem Network.
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Valid Entry accunet cbc
dmi-mos i800 inwats lds megacom
mega800
multiquest
operator outwats-bnd
public-ntwrk sddn
sdn
sub-operator tandem tie wats-max-bnd
Usage
ACCUNET Switched Digital Service -- part of ACI (AT&T Communications ISDN) phase 2.
Call-by-Call service -- provides different dial plans for different services on an ISDN trunk group. Indicates this trunk group is used by the Call-By-Call Service Selection feature.
Digital multiplexed interface -- message-oriented signaling.
International 800 Service -- A subscriber can receive international calls without a charge to the call originating party.
INWATS -- provides OUTWATS-like pricing and service for incoming calls.
Long-Distance Service -- part of ACI (AT&T Communications ISDN) phase 2.
MEGACOM Service -- an AT&T communications service that provides unbanded long-distance services using special access (switch to 4ESS switch) from an AT&T communications node.
MEGACOM 800 Service -- an AT&T communications service that provides unbanded 800 service using special access (4ESS switch to switch) from an AT&T communications node.
AT&T MULTIQUEST Telecommunications Service -- dial 700 service. A terminatinguser's service that supports interactive voice service between callers at switchedaccess locations and service provides directly connected to the AT&T Switched Network (ASN).
Network Operator -- provides access to the network operator.
OUTWATS Band -- WATS is a voice-grade service providing both voice and low speed data transmission capabilities from the user location to defined service areas referred to as bands; the widest band is 5.
Public network calls -- It is the equivalent of CO (outgoing), DID, or DIOD trunk groups. If Service Type is public-ntwrk, Dial Access can be enabled.
Software Defined Data Network -- provides a virtual private line connectivity via the AT&T switched network (4ESS switches). Services include voice, data, and video applications. These services complement the SDN service. Do not use for DCS with Rerouting.
Software Defined Network (SDN) -- an AT&T communications offering that provides a virtual private network using the public switched network. SDN can carry voice and data between customer locations as well as off-net locations.
Presubscribed Common Carrier Operator -- provides access to the presubscribed common carrier operator.
Tandem tie trunks integral to an ET.
Tie trunks -- general purpose.
Maximum Banded Wats -- a WATS-like offering for which a user's calls are billed at the highest WATS band subscribed to by users.
Related links Facility Type on page 630
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ISDN Trunk Group
Signaling Group
Available only if the Carrier Medium is H.323 or SIP, and the Member Assignment Method is auto.
Valid Entry digits or blank
Usage
Assigned H.323 signaling group number. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Related links Carrier Medium on page 505 Member Assignment Method on page 511
TAC
The trunk access code (TAC) that must be dialed to access the trunk group. A different TAC must be assigned to each trunk group. CDR reports use the TAC to identify each trunk group. The characters "*" and "# " can be used as the first character in a TAC. Accepts a one- to four-digit number.
TestCall BCC
The Bearer Capability Code (BCC) used for the ISDN test call.
Valid Entry 0 1 2 4
Usage Voice Mode 1 Mode 2 Asynchronous Mode 0
Testcall ITC
Controls the encoding of the Information Transfer Capability (ITC) codepoint of the bearer capability Information Element (IE) in the SETUP message when generating an ISDN test call.
Note:
The ISDN Testcall feature has no routing, so a testcall is never blocked due to an incompatible ITC.
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Valid Entry rest unre
Usage Restricted Unrestricted
ISDN Trunk Group: page 1
Testcall Service
The call-by-call selection for an ISDN test call. Available only if the Service Type is cbc. Unavailable for Facility Type 0 (feature), 1 (service), or 3 (outgoing) that is defined by users.
Valid Entry access accunet cbc
dmi-mos i800 inwats lds megacom
mega800
multiquest
operator outwats-bnd
public-ntwrk sdn
Usage
A tie trunk giving access to an Electronic Tandem Network.
ACCUNET Switched Digital Service -- part of ACI (AT&T Communications ISDN) phase 2.
Call-by-Call service -- provides different dial plans for different services on an ISDN trunk group. Indicates this trunk group is used by the Call-By-Call Service Selection feature.
Digital multiplexed interface -- message-oriented signaling.
International 800 Service -- allows a subscriber to receive international calls without a charge to the call originating party.
INWATS -- provides OUTWATS-like pricing and service for incoming calls.
Long-Distance Service -- part of ACI (AT&T Communications ISDN) phase 2.
MEGACOM Service -- an AT&T communications service that provides unbanded long-distance services using special access (switch to 4ESS switch) from an AT&T communications node.
MEGACOM 800 Service -- an AT&T communications service that provides unbanded 800 service using special access (4ESS switch to switch) from an AT&T communications node.
AT&T MULTIQUEST Telecommunications Service -- dial 700 service. A terminatinguser's service that supports interactive voice service between callers at switchedaccess locations and service provides directly connected to the AT&T Switched Network (ASN).
Network Operator -- provides access to the network operator.
OUTWATS Band -- WATS is a voice-grade service providing both voice and low speed data transmission capabilities from the user location to defined service areas referred to as bands; the widest band is 5.
Public network calls -- It is the equivalent of CO (outgoing), DID, or DIOD trunk groups. If Service Type is public-ntwrk, Dial Access can be enabled.
Software Defined Network (SDN) -- an AT&T communications offering that provides a virtual private network using the public switched network. SDN can carry voice and data between customer locations as well as off-net locations.
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ISDN Trunk Group
Valid Entry sub-operator
tandem tie wats-max-bnd
Usage
Presubscribed Common Carrier Operator -- provides access to the presubscribed common carrier operator.
Tandem tie trunks integral to an ET.
Tie trunks -- general purpose.
Maximum Banded Wats -- a WATS-like offering for which a user's calls are billed at the highest WATS band subscribed to by users.
Related links Service Type on page 512 Facility Type on page 630
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
Usage Alloc
If enabled, allocates service provided by the trunk group and enhances Network Call Redirection. Available only if the ISDN Trunk Group Service Type is cbc.
Related links Service Type on page 512
ISDN Trunk Group: page 2
Administer Timers
Enables or disables administration of timers on this trunk group. The default for the ISDN trunk group type is disabled. All other trunk group types are enabled by default. Available for all trunk group types except cpe, h.323, and sip. Related links Group Type on page 508
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ISDN Trunk Group: page 2
Answer Supervision Timeout
Valid Entry 0 to 250
Usage
The number of seconds Communication Manager waits before it acts as though answer supervision has been received from the far-end. During a cut-through operation, timing begins after each outgoing digit is sent and timing ceases after the far-end sends answer supervision. On senderized operation, the timer begins after the last digit collected is sent.
Note:
This field's setting does not override answer supervision sent from the network or from DS1 port circuit timers.
Related links Receive Answer Supervision on page 649 Administer Timers on page 516
Bit Rate
Valid Entry 300 1200 2400 4600 9600 19200
Usage
The baud rate used by pooled modems. The speed of the fastest modem that uses this ISDN trunk group.
Charge Advice
Determines how to accumulate and access charge information about a call. Requires that CDR Reports be enabled before changing this field from its default of none. Receiving Advice of Charge during the call affects system performance because of the increased ISDN message activity on the signaling channel, which might reduce the maximum call capacity.
Valid Entry none automatic
Usage The system does not to collect Advice of Charge information for this trunk group. The public network sends Advice of Charge information automatically.
Table continues...
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ISDN Trunk Group
Valid Entry end-on-request
during-on-request
Usage
Communication Manager requests charge information with each call, and you receive only the final call charge.
Communication Manager requests charge information with each call, and charges display during and at the end of a call.
Codeset to Send Display
Defines the codeset for sending the information element for display. The value depends on the type of server or switch used for the connection.
Valid Entry 0 6 7
Usage CCITT (non-Communication Manager equipment). Any other than CCITT or System 85 R2V4, 4E11. System 85 R2V4, 4E11.
Codeset to Send National IEs
The codeset for sending the information element (IE) for national IEs. National IEs include all IEs previously sent only in code set 6 (such as DCS IE). Now these national IEs, including Traveling Class Marks (TCMs) and Lookahead Interflow (LAI), can be sent in code set 6 or 7. The value depends on the type of server/switch to which the user is connected.
Valid Entry 6 7
Usage Other types. System 85 R2V4, 4E11, or newer Avaya S8XXX Server types.
Note:
A Traveling Class Mark (that is, the user's FRL or the user's trunk group FRL) is passed between tandem nodes in an ETN in the setup message only when the Service Type is tandem. It then is used by the distant tandem switch to permit access to facilities consistent with the originating user's privileges.
CONNECT Reliable When Call Leaves ISDN
Available only if Group Type is ISDN. The value tells the Communication Manager server whether a CONNECT received on an outgoing call that is not end-to-end ISDN is a reliable indication that the far end has answered the call.
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Valid entry y n
ISDN Trunk Group: page 2
Usage The CONNECT is a reliable answer. The CONNECT message is an unreliable answer. The message can be the result of a timer expiring. If the call was originated by a Call Center adjunct, a Call Classifier can be used instead to determine whether the call has been answered. The default value is n.
Digit Handling (in/out)
Valid Entry enbloc/enbloc enbloc/overlap overlap/enbloc overlap/overlap
Usage
Defines whether overlap receiving and overlap sending features are enabled. enbloc disables overlap receiving and overlap sending. The first field value indicates digit receiving and the second value indicates digit sending.
Digital Loss Group
Valid Entry 1 to 19
Usage
Determines which administered two-party row in the loss plan applies to this trunk group if the call is carried over a digital signaling port in the trunk group. If values other than 18 or between 11 and 15 are administered, the system displays a warning message indicating that the loss group may not be appropriate for this trunk group.
Disconnect Supervision-Out
Indicates whether Communication Manager receives disconnect supervision for outgoing calls over this trunk group. Available for outgoing or two-way trunk groups.
Valid Entry y
n
Usage
Allows trunk-to-trunk transfers involving trunks in this group. The far-end sends a release signal when the called party releases an outgoing call, and the far-end is responsible for releasing the trunk. Enhances Network Call Redirection. Available only if Answer Supervision Timeout is 0 and Receive Answer Supervision is enabled.
The far-end server or switch does not provide a release signal, the hardware cannot recognize a release signal, or timers are preferred for disconnect supervision on outgoing calls. Prevents trunk-to-trunk transfers involving trunks in this group.
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ISDN Trunk Group
Caution: Verify that the far-end server or switch provides answer supervision and disconnect supervision. Most public networks do not provide disconnect supervision over analog trunks. Check with the network services provider. Related links Direction on page 507 Answer Supervision Timeout on page 517 Receive Answer Supervision on page 649
Duplex
Note:
Even if the trunk group supports full-duplex transmission, other equipment in a circuit might not.
Valid Entry full
half
Usage
Provides simultaneous two-way transmission, which is most efficient. Recommended in most cases.
Supports only one transmission direction at a time.
Group Type
Displays the type of trunk group.
Related links Group Type on page 508 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
Max Message Size to Send
Valid Entry 128 244 256 260
Usage
The maximum number of bytes of an ISDN message for Communication Manager.
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ISDN Trunk Group: page 2
The following table indicates the expected ISDN-PRI message size from several Lucent Technologies and Avaya Inc. products.
Products 4ESS (4E11) 4ESS (4E13) 4ESS (4E14) 5ESS (5E4) 5ESS (5E5) 5ESS (5E6) System 75 (all) System 85 (R2V4) System 85 (R2V5) System 85 (R2V6)
Message Length (octets) Received 256 256 256 244 244 244 260 128 260 260
Supplementary Service Protocol
The supplementary service protocol to use for services over this trunk group. Supplementary service protocols are mutually exclusive.
Valid Entry a b c d
e f g
Usage National. ISO/ETSI QSIG Private Network. Also used for SBS signaling trunks. ETSI public network. European Computer Manufacturer's Association (ECMA) QSIG private network (supports only Name Identification and Additional Network Feature Transit Counter (ANF-TC)). DCS with Rerouting. Do not use with Service Type of dmi-mos or sddn. ISDN Feature Plus Public network feature plus signaling. ANSI.
Related links SBS on page 530 Local Country Code on page 322 International Access Code on page 323 Group Type on page 508 Carrier Medium on page 505
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ISDN Trunk Group
Synchronization
Caution: Do not change this field without the assistance of Avaya or the network service provider.
Valid Entry async sync
Usage
Determines whether the trunk group uses synchronous or asynchronous communications.
Trunk Hunt
Defines the trunk hunt search order. Communication Manager performs a trunk hunt when searching for available channels within a facility in an ISDN trunk group. The search can be administered per ISDN-PRI trunk group, but it infers the direction of search within all ISDN-PRI facilities (or portions of those facilities) administered within the trunk group.
Valid Entry ascend cyclical
descend
Usage
Enables a linear trunk hunt search from the lowest to highest numbered channels. All trunks within an ISDN trunk group are selected without regard to the order in which trunks are administered within the trunk group.
Enables a circular trunk hunt based on the sequence the trunks were administered within the trunk group. When using ISDN-BRI interfaces, only cyclical is supported. The cyclical option cannot be set if the trunk group using ISDN-PRI interfaces is to be used for Wideband operations.
Enables a linear trunk hunt search from the highest to lowest numbered channels. All trunks within an ISDN trunk group are selected without regard to the order in which trunks are administered within the trunk group.
Related links Wideband Support on page 534
Trunk Type
The type of trunk. Related links
Trunk Type (in/out) on page 918
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ISDN Trunk Group: page 3
ISDN Trunk Group: page 3
ACA Assignment
Indicates whether Automatic Circuit Assurance (ACA) measurements are taken for this trunk group.
Apply Local Ringback
Enables or disables local ringback tone to the caller. If enabled, local ringback is removed when the call is connected. Available only if the Carrier Medium is PRI_BRI.
Related links Carrier Medium on page 505
BSR Reply-best DISC Cause Value
Servers running Communication Manager that are polled as resources in a Best Service Routing application return data to the polling server in the ISDN DISC message. Since some cause values do not work over some networks, this field sets the cause value that the server returns in response to a BSR status poll. If this field is set incorrectly, incoming status poll calls over this trunk group are dropped before any data is returned to the polling server or switch.
Available only if UUI IE Treatment is shared.
Valid Entry 31 17 16
Usage Normal-unspecified. This value is almost always used. User-busy Normal-call-clearing.
Caution:
In most cases, this field is set to the appropriate value during installation. Do not change this field without the assistance of Avaya or your network service provider.
Related links UUI IE Treatment on page 534
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ISDN Trunk Group
Charge Conversion
Available only for outgoing or two-way CO, DIOD, FX, and WATS trunk groups. Also available for isdn trunk groups, if the Charge Advice field is set to a value other than none.
Valid Entry 1 to 64 500
Usage
Communication Manager multiplies the number of charge units by the value of this field and displays it as a currency amount. Without a value in this field, Communication Manager displays the number of charge units without converting it to currency.
Related links Direction on page 507 Trunk Direction on page 649
Currency Symbol
The telephone displays the symbol before the charge amount. Accepts from one to three characters. Leading and embedded spaces count as characters.
Available only for outgoing or two-way CO, DIOD, FX, and WATS trunk groups. Also available for ISDN trunk groups, if the Charge Advice field is set to a value other than none.
Related links Direction on page 507 Trunk Direction on page 649
Data Restriction
If y, whisper page is denied on this trunk. Enter y to prevent features from generating tones on a data call that would cause erroneous data transmission.
Decimal Point
The appropriate representation for a decimal point as telephone displays it. Available only with outgoing or two-way CO, DIOD, FX, and WATS trunk groups. Also available for isdn trunk groups, if the Charge Advice field is set to a value other than none.
Note:
If the received charge contains no decimals, no decimal point is displayed (that is, the administered decimal point is ignored for charge information received with no decimals). On a QSIG trunk group, unlike other trunk groups, the Decimal Point field is independent whether
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the calling display shows a decimal point. Instead, it tells what symbol should be displayed if the QSIG AOC received has a 1/10 or 1/100 or 1/1000 Multiplier.
Valid Entry comma
period
none
Usage
If the received charge contains decimals, the charge is displayed at the calling endpoint's display with a comma as the decimal point. Divides the charge value by 100.
This is the default. If the received charge contains decimals, the charge is displayed at the calling endpoint's display with a period as the decimal point. Divides the charge value by 100.
No decimal point is displayed.
Related links Direction on page 507 Charge Advice on page 506
DS1 Echo Cancellation
Enables or disables echo cancellation on a per port basis. If enabled, reduces voice call echo. Note: Changes to the DS1 Echo Cancellation field is incomplete until one of the following occurs: · Port is busied-out or released. · Trunk group is busied-out or released. · SAT command test trunk group is performed. · Periodic maintenance runs.
DSN Term
Enables or disables the trunk group as a DSN termination telephone. The default is disabled.
Format
Specifies the encoding of Numbering Plan Indicator for identification purposes in the Calling Number and/or Connected Number IEs, and in the QSIG Party Number. If you have used Special Application SA9118 to generate international numbers in a Service Pack for Communication Manager Releases 5.2.1 or 6.0.1, you must set the Format field to intl-pub for the appropriate ISDN trunk group(s) after upgrading a Communication Manager server to Release 6.2 or later, because there is no upgrade path for intl-pub entry in Communication Manager 6.2.
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ISDN Trunk Group
The Format field is available only if the Send Calling Number or Send Connected Number field is enabled or restricted.
Note:
The entries private and unk-pvt are overridden, and a public number is generated, if the called number for the outgoing call over the trunk group is encoded using E.164. The numbering plan for the called number is determined by the Call Type field from the ARS or AAR Analysis, or the Numbering Format field from the outgoing Route Pattern.
Valid Entry intl-pub natl-pub private pub-unk unk-pvt
Usage
Indicates the Numbering Plan Indicator is E.164. International (public number that includes city or area code and the country code).
Indicates the Numbering Plan Indicator is E.164. National (public number that includes city or area code, but excludes the country code).
Indicates the Numbering Plan Indicator is Private. Based on the network level of the ISDN Parameters field on the Feature-related System Parameters screen.
Indicates the Numbering Plan Indicator is E.164. Unknown (public number with no details of the digits included).
Indicates the Numbering Plan Indicator is Private. Unknown (private number with no details about the digits included).
Related links Send Calling Number on page 530 Send Connected Number on page 531 Numbering-Private Format on page 633
Maintenance Tests
Enables or disables hourly maintenance tests on this trunk group. Available only for aplt, isdn, sip, or tie trunk groups. Related links Group Type on page 508
Maximum Size of UUI IE Contents
Available only if the UUI IE Treatment is shared.
Valid Entry 32 to 128
Usage The maximum number of bytes of user information that the network supports.
Related links UUI IE Treatment on page 534
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ISDN Trunk Group: page 3
Measured
Indicates if the system transmits data for this trunk group to the Call Management System.
Valid Entry internal
external both
none
Usage Sends the data to the Basic Call Management System (BCMS), the VuStats data display, or both.
Available only if BCMS (Basic) or VuStats is enabled for the system. Sends the data to the CMS. Collects data internally and sends it to the CMS.
Available only if BCMS (Basic) or VuStats is enabled for the system. Trunk group measurement reports are not required.
Related links VuStats on page 847 BCMS (Basic) on page 842
Modify Tandem Calling Number
Available with outgoing or two-way trunks when the Carrier Medium field is set to PRI/BRI or H.323, and the Send Calling Number field is set to enabled or restricted.
Valid Entry natl-intl-prefix
tandem-cpn-form
no
Usage
Adds the national or international prefixes from the Locations Parameters screen when the calling party number is appropriate.
Modifies the calling party number IE in the previously administered format specified for the Tandem Calling Party Number.
Does not modify the calling party number.
Related links Group Type on page 508 Direction on page 507 Carrier Medium on page 505 Send Calling Number on page 530 Modify Tandem Calling Number on page 944
NCA-TSC Trunk Member
The trunk member number whose D-channel is used to route tandem NCA-TSCs or QSIG CISCs.
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ISDN Trunk Group
Network Call Redirection
Whenever the Supplementary Service Protocol is changed, this field resets to none to prevent an inadvertent incorrect value.
Available only if ISDN Network Call Redirection is enabled for the system and the Supplementary Service Protocol is a, c, or g.
Valid entry deflect ANSI-transfer Nortel-transfer telcordia-tbct
Usage Allows Network Call Deflection. Allows Network Call Transfer for MCI DEX 600 ISDN trunks. Allows Network Call Transfer for MCI DMS 250 switches. Allows Network Call Transfer for Lucent 5ESS or Nortel DMS100 switches.
Related links ISDN-PRI on page 836 ISDN/SIP Network Call Redirection on page 836 Supplementary Service Protocol on page 521
Network (Japan) Needs Connect Before Disconnect
Sends an ISDN Connect message just prior to the disconnect message.
Outgoing Channel ID Encoding
Determines whether to encode the Channel ID IE as preferred or exclusive. Available only if the Group Type is isdn and the Service Type is anything except dmi-mos or sddn. Blank is an invalid entry.
Valid Entry preferred exclusive
Usage Default if Used for DCS is disabled or unavailable. Default if Used for DCS is enabled.
Related links Group Type on page 508 Service Type on page 512
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Path Replacement Method
Available only if Basic Call Setup and Supplementary Services with Rerouting are enabled for the system and when the Supplementary Service Protocol is either b or e and the Group Type is isdn.
Valid Entry better-route
always
Usage
Uses the most economical route; for example, the reconfigured call does not use the same trunk group as the original call.
Always reconfigures the call regardless of the trunk group used.
Related links ISDN-PRI on page 836 ISDN-BRI Trunks on page 835 Basic Call Setup on page 848 Supplementary Services with Rerouting on page 849 Service Type on page 512 Group Type on page 508
Replace Restricted Numbers
Indicates whether to replace restricted numbers with administrable strings for incoming and outgoing calls assigned to the specified trunk group. If enabled, the display is replaced regardless of the service type of the trunk. Applies to BRI, PRI, H.323, and SIP trunks. Available only if the Group Type is isdn or sip.
Related links Group Type on page 508
Replace Unavailable Numbers
If enabled, replaces unavailable numbers with administrable strings for incoming and outgoing calls assigned to the specified trunk group. The display is replaced regardless of the service type of the trunk. Applies to BRI/PRI, H.323, and SIP trunks. Also applies to analog trunks if Analog Trunk Incoming Call ID is enabled and Receive Analog Incoming Call ID is set to any value except disabled. Available only if the Group Type is isdn or sip.
Related links Group Type on page 508 Analog Trunk Incoming Call ID on page 828 Receive Analog Incoming Call ID on page 947
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ISDN Trunk Group
SBS
Enables or disables Separation of Bearer and Signaling (SBS) for the trunk group.
Available only if the Local Country Code and International Access Code are administered for the system and when the Supplementary Service Protocol is b, the Group Type is isdn, the Carrier Medium is H.323, and Dial Access is disabled.
Related links Local Country Code on page 322 International Access Code on page 323 Group Type on page 508 Supplementary Service Protocol on page 521 Carrier Medium on page 505
Send Called/Busy/Connected Number
Specifies if the dialed number, whether called (ringing), busy (busy tone), or connected (answered) is sent on incoming or tandemed ISDN calls.
Available only if QSIG Value-Added is enabled for the trunk group.
Valid Entry y
n r
Usage
The dialed number is sent on incoming or tandemed ISDN calls. This field must be enabled in order for the Calling Party Number of an incoming ISDN call to display at the transferred-to station after a QSIG transfer operation. If enabled, the Numbering - Public/Unknown Format is accessed to construct the actual number sent, or the Numbering - Private Format is used.
Disables the sending of the dialed number on incoming or tandemed ISDN calls.
Restricted. The connected number is sent "presentation restricted".
Related links Numbering -- Public/Unknown Format on page 636 QSIG Value-Added on page 537 Numbering-Private Format on page 633
Send Calling Number
Specifies whether the calling party's number is sent on outgoing or tandemed ISDN calls.
Note: The Numbering - Public/Unknown Format can override the Send Calling Number administration
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ISDN Trunk Group: page 3
Valid Entry y
n
r
Usage
The calling party's number is sent on outgoing or tandemed ISDN calls. If enabled, the Numbering - Public/Unknown Format is accessed to construct the actual number sent, or the Numbering - Private Format is used.
Disables the sending of the calling party's number on outgoing or tandemed ISDN calls. If disabled, an incoming number is not tandemed out again. This applies to all Supplementary Service Protocols.
Restricted. The calling number is sent "presentation restricted". If set to restricted, an incoming number is marked restricted when it is tandemed out again. This applies to all Supplementary Service Protocols.
Related links Numbering-Private Format on page 633 Incoming Number Format on page 100 Numbering -- Public/Unknown Format on page 636
Send Codeset 6/7 LAI IE
If enabled, the ISDN trunk transmits information in Codeset 6/7.
If the UUI IE Treatment is shared, then this field should be disabled. Otherwise, the same information will be sent twice and might exceed the message size.
Related links UUI IE Treatment on page 534
Send Connected Number
Specifies if the connected party's number is sent on incoming or tandemed ISDN calls. Available only if QSIG Value-Added is disabled for the trunk group.
Valid Entry y
n r
Usage
The connected party's number is sent on outgoing or tandemed ISDN calls. If enabled, the Numbering - Public/Unknown Format is accessed to construct the actual number sent, or the Numbering - Private Format is used. This field must be enabled for the Calling Party Number of an incoming ISDN call to display at the transferred-to station after a QSIG transfer operation.
Disables the sending of the connected party's number on outgoing or tandemed ISDN calls. If disabled, an incoming number is not tandemed out again. This applies to all Supplementary Service Protocols.
Restricted. The connected number is sent "presentation restricted". If this field is set to r, an incoming number is marked restricted when it is tandemed out again. This applies to all Supplementary Service Protocols.
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ISDN Trunk Group
Note:
The AT&T Switched Network Protocol does not support restricted displays of connected numbers. Therefore, if you administer the 1a country-protocol/ protocol-version combination for the DS1 Circuit Pack, you should not administer the Send Connected Number as restricted, as this causes display problems. The Numbering - Public/Unknown Format overrides the Send Connected Number administration for any administrable block of extensions.
Related links Numbering-Private Format on page 633 QSIG Value-Added on page 537 Numbering -- Public/Unknown Format on page 636
Send Name
Specifies whether the calling/connected/called/busy party's administered name, or the name on a redirected call, is sent to the network on outgoing/incoming calls.
Note:
If name information is not administered for the calling station or the connected/called/busy station, the system sends the extension number in place of the name.
Valid Entry y n
r
Usage
The calling/connected/called/busy party's administered name and the name on a redirected call are sent on outgoing/incoming calls. An entry of y or n is required if the Supplementary Service Protocol is e (DCS with Rerouting).
Disables the sending of the calling/connected/called/busy party's administered name and the name on a redirected call on outgoing/incoming calls. An entry of y or n is required if the Supplementary Service Protocol is e (DCS with Rerouting). With an entry of n, an incoming name is not tandemed out again if the Supplementary Service Protocol field is any value other than b (QSIG).
Restricted. The calling/connected/called/busy party's administered name is sent "presentation restricted" on outgoing/incoming calls. The name on a redirected call is not sent. This value is valid only if the Supplementary Service Protocol is a (national supplementary service), b (for called/ busy only), or d (for the QSIG Global Networking Supplementary Service Protocol). With an entry of r, an incoming name is marked restricted when it is tandemed out again. However, if the Supplementary Service Protocol is b (QSIG), then an incoming name is passed on unchanged and Send Name value is ignored.
Related links Supplementary Service Protocol on page 521
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ISDN Trunk Group: page 3
Send UCID
If you enable the Send UCID field, the trunk transmits Universal Call IDs (UCIDs). This field is available only if the UUU IE Treatment field is set to Shared.
Note:
If the incoming trunk has two UCIDs in the UUI, the outgoing trunk sends the UUI with both UCIDs to the far-end switch.
Send UUI IE
If enabled, sends UUI information on a per trunk group basis.
Show ANSWERED BY on Display
Available only for isdn pri/bri, isdn h.323, and sip trunk groups.
Valid Entry y
n
Usage
The words "ANSWERED BY" display in addition to the connected telephone number on calls over this trunk. This is the default.
Note:
Based on display language settings for stations, "ANSWERED BY" is translated into and displayed in the appropriate language.
Only the connected telephone number displays. This might be preferred when outgoing calls are over a trunk that might be redirected.
Suppress # Outpulsing
Indicates whether or not to suppress the final "#" in cases where the system would normally outpulse it. Used if end-to-end signaling begins with (and includes) "#". This field should be enabled when the local telephone company central office or any other facility treats "#" as an error.
US NI Delayed Calling Name Update
For digital display stations, if calling name information is received after the incoming call has been delivered to the terminating telephone, enables or disables a display update.
For analog display stations, Communication Manager is enhanced to wait for the FACILITY message (that contains the name information) before delivering an incoming call to the receiver.
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ISDN Trunk Group
You can administer the time duration for which Communication Manager waits for the FACILITY message.
Note: BRI trunks do not support display updates. Available only if the Carrier Medium is either PRI/ BRI or ATM, and the Supplementary Service Protocol is a. Related links Supplementary Service Protocol on page 521 Carrier Medium on page 505 ISDN-PRI on page 836
UUI IE Treatment
Specifies whether the user Information Element (IE) is shared.
Valid Entry shared service-provider
Usage The trunk is connected to the server. Service provider functionality is required.
Wideband Support
Note:
This feature is not supported on the DS1 interfaces on branch gateways (G700).
Enables or disables wideband switching on this trunk group. Only trunk members from TN464C or later circuit packs can use wideband switching.
Available only if Wideband Switching is enabled for the system.
Note:
Wideband trunk calls are treated as a single trunk call when Automatic Circuit Assurance (ACA) measurements are taken. This way, if an ACA referral call is generated (for short or long holding time), the wideband call only triggers a single referral call using the lowest Bchannel trunk member associated with the wideband channel.
Related links Wideband Switching on page 842
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QSIG Trunk Group Options
QSIG Trunk Group Options
The fields on this screen appear only when Group Type is isdn and Supplementary Service Protocol is b. Related links Group Type on page 508 Supplementary Service Protocol on page 521 Trunk Group on page 901
Character Set for QSIG Name
Use this field to configure the character set for transmission and display of the QSIG name data. This field is available only if:
· Group Type is isdn. · Supplementary Service Protocol is b. · Display Character Set is Roman.
Valid Entry eurofont
iso-8859-1
Usage
The Roman Eurofont character set. Eurofont is the default setting.
Note:
Avaya BRI stations support only ASCII characters. A BRI station does not correctly display non-ASCII characters, such as Eurofont or Kanafont.
The system converts all data or characters transmitted over a QSIG trunk from Eurofont to ISO 8859-1. Eurofont is an Avaya proprietary encoding character set.
Note:
ISO 8859-1, also known as ISO/IEC 8859-1 or Latin-1, is part 1 of ISO/IEC 8859, a standard character encoding defined by ISO. ISO 8859-1 encodes Latin alphabet no. 1, which consists of 191 characters from the Latin script. Each character is encoded as a single 8-bit code value.
Related links Group Type on page 508 Supplementary Service Protocol on page 521 Display Character Set on page 821
Diversion by Reroute
Use this field to activate or deactivate the Diversion by Reroute feature. The default value of this field is y.
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ISDN Trunk Group
This field is available only if: · Group Type is ISDN. · Supplementary Service Protocol is b.
If the value of this field is y, Communication Manager selects the optimum path and routes the call on that path to the destination number. If the value of this field is n, Communication Manager does not initiate a Diversion or Reroute request over the QSIG trunk. Communication Manager also rejects any Diversion or Reroute request that Communication Manager receives over the QSIG trunk. Related links Supplementary Service Protocol on page 521 Group Type on page 508
Path Replacement
Enables or disables the Path Replacement feature. The default is enabled. If disabled, Communication Manager does not originate a Path Replacement request over that trunk group, and rejects any Path Replacement request it receives over that trunk group. Available only if the Group Type is isdn and the Supplementary Service Protocol is b.
Related links Group Type on page 508 Supplementary Service Protocol on page 521
Path Replacement Method
Available only if the Group Type is ISDN, the Supplementary Service Protocol is b or e, and Supplementary Services with Rerouting or DCS with Rerouting is enabled for the system. Unavailable if Path Replacement with Retention is enabled.
Valid Entry always
BR (better route)
Usage
Use any QSIG (SSB) trunk group as the replacement trunk group. A new call is always originated, even when the original trunk group is determined to be the replacement trunk group.
Route pattern preferences help determine trunk group path replacement. The original trunk group is retained if Path Replacement with Retention is enabled. Path replacement fails (and the original trunk group is retained) if Path Replacement with Retention is disabled.
Related links Group Type on page 508 Supplementary Service Protocol on page 521 Supplementary Services with Rerouting on page 849
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DCS with Rerouting on page 832 Path Replacement with Retention on page 537
QSIG Trunk Group Options
Path Replacement with Retention
Available only if the Group Type is ISDN, the Supplementary Service Protocol is b or e, and Supplementary Services with Rerouting or DCS with Rerouting is enabled for the system.
Valid Entry y n
Usage
Retains the original trunk group.
Provides path replacement according to settings for the Path Replacement Method.
Related links Supplementary Service Protocol on page 521 Supplementary Services with Rerouting on page 849 DCS with Rerouting on page 832 Group Type on page 508
QSIG Value-Added
Enables or disables QSIG-VALU services. Available only if Value-Added (VALU) is enabled for the system and Supplementary Services Protocol is enabled for the trunk.
Related links Value Added (VALU) on page 849 Supplementary Service Protocol on page 521
QSIG-Value Coverage Encoding
The encoding method used to encode DL1, DL2, and DL3 extensions. Available only if the Group Type is isdn, Supplementary Service Protocol is b, and QSIG Value-Added is enabled for the system.
Valid Entry proprietary
standard
Usage
Communication Manager sends extension information in the normal manner. This is the default.
In addition to normal extension information, Communication Manager sends the data part (as null) of the extension.
Related links Value Added (VALU) on page 849
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ISDN Trunk Group
Group Type on page 508 Supplementary Service Protocol on page 521
SBS
Enables or disables Separation of Bearer and Signaling (SBS) for the trunk group.
Available only if the Local Country Code and International Access Code are administered for the system and when the Supplementary Service Protocol is b, the Group Type is isdn, the Carrier Medium is H.323, and Dial Access is disabled.
Related links Local Country Code on page 322 International Access Code on page 323 Group Type on page 508 Supplementary Service Protocol on page 521 Carrier Medium on page 505
SIP Reference Trunk Group
The system displays SIP Reference Trunk Group only when the Group Type field is isdn, the Carrier Medium field is H.323, and the Supplementary Service Protocol field is b.
Valid Entry 1 to 2000
blank
Usage
Assigns a number for the SIP trunk group. If trunk members are already assigned to this trunk group, you must not change the value of SIP Reference trunk Group field. If you change the value of this field, the system displays an error message and prompts you to remove all assigned members before enabling Q-SIP.
No SIP trunk group is assigned. By default, the value is blank.
Note:
For the maximum allowed entry per server type, see Avaya Aura® Communication Manager System Capacities Table.
TSC Method for Auto Callback
Controls the signaling connection method for the QSIG Temporary Signaling Connections (TSC) when Communication Manager is the terminating or the outgoing gateway PINX.
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ISDN Trunk Group: Administrable Timers
Valid Entry drop-if-possible always-retain
Usage
QSIG Temporary Signaling Connections are released.
QSIG Temporary Signaling Connections are always retained for successful call completion activation.
ISDN Trunk Group: Administrable Timers
This screen displays only when Administer Timers is enabled. This screen does not display ISDN trunk group information for trunks of Group Type cpe or sip.
Note: If the ISDN trunk group has a Carrier Medium value of H.323, or if the trunk group has BRI members, then this page cannot be administered. In these cases, an error message displays when you attempt to submit the screen.
Caution: Customers: Do not change fields on this page without assistance from Avaya or your network service provider. Related links Group Type on page 508 Administer Timers on page 516 Carrier Medium on page 505
Programmed Dial Pause (msec)
Valid Entry
100 to 25500 in increments of 100
Usage
The exact duration of the pause used during abbreviated dialing, ARS outpulsing, and terminal dialing operations. This timer is administrable for all outgoing and twoway trunk groups. This timer works with the TN464B (or later), TN767, TN458, TN2140, and TN2242 tie circuit packs. All central office (CO) circuit packs that accept administrable timers accept this timer.
END TO END SIGNALING
Pause (msec)
Available only if the Trunk Type is blank.
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ISDN Trunk Group
Valid Entry
20 to 2550 in increments of 10
Usage
The interval (pause) between DTMF tones sent from a hybrid telephone. All CO, DIOD, and tie circuit packs that accept administrable timers accept this timer. However, this timer is sent only to the following circuit packs: TN464B (or later), TN767, TN436B, TN459B, TN2146, TN2199, and TN2242, and TN429 and TN2184 ports in a DID trunk group.
Related links Trunk Type on page 936
Tone (msec)
Available only if the Trunk Type is blank.
Valid Entry
20 to 2550 in increments of 10
Usage
The duration of the DTMF tone sent when a button on a hybrid telephone is pressed. All CO, DIOD, and Tie circuit packs that accept administrable timers accept this timer. This timer is also sent to the following circuit packs: TN464B (or later), TN767, TN436B, TN459B, TN2146, TN2199, TN429, TN2184 ports in a DID trunk group.
Related links Trunk Type on page 936
Shared UUI Feature Priorities
The fields in this page show the priorities for each type of information to be forwarded in the Shared UUI. The system displays this page only on the ISDN trunk group screen when all of the following conditions are met:
· The UUI IE Treatment is shared. · The Supplementary Service Protocol is set to anything except b. Changing the priorities in this screen might affect whether certain information will be sent. Related links Supplementary Service Protocol on page 521 UUI IE Treatment on page 534
ASAI
User information from Adjunct/Switch Applications Interface (ASAI).
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Valid Entry 1 to 7 blank
Shared UUI Feature Priorities
Usage Level of priority, with 1 being the highest. Default priority is 1. This field's information is not forwarded.
Collected Digits
Digits collected from caller (not including dial-ahead digits).
Valid Entry 1 to 7 blank
Usage Level of priority, with 1 being the highest. Default priority is 5. This field's information is not forwarded.
Held Call UCID
The unique tag for the last call that was put on hold by the Automatic Call Distribution (ACD) agent placing this call to another system. This Universal Call ID (UCID) can be used to identify the original or parent call that may eventually be placed into conference or transferred to the other system. This element is required for cradle-to-grave tracking with Avaya IQ release 5.0 and later.
Valid Entry 1 to 7
blank
Usage
Level of priority, with 1 being the highest. The UCID included in the element with default priority 2 is the tag for a new call placed by the agent while the original call is on hold. Default priority is 7.
This field's information is not forwarded.
In-VDN Time
Number of seconds the call has spent in vector processing.
Valid Entry 1 to 7 blank
Usage Level of priority, with 1 being the highest. Default priority is 3. This field's information is not forwarded.
Other LAI Information
Includes the time stamp of when the call entered the current queue, the call's priority level in its current queue, and the type of interflow.
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ISDN Trunk Group
Valid Entry 1 to 7 blank
Usage Level of priority, with 1 being the highest. Default priority is 6. This field's information is not forwarded.
Universal Call ID
A unique tag that identifies the call that this message is being sent for and the other information included in the User-User-Information (UUI).
Valid Entry 1 to 7 blank
Usage Level of priority, with 1 being the highest. Default priority is 2. This field's information is not forwarded.
VDN Name
Name of the active VDN (also called LAI DNIS).
Valid Entry 1 to 7 blank
Usage Level of priority, with 1 being the highest. Default priority is 4. This field's information is not forwarded.
CBC Service Trunk Group Allocation Plan Assignment Schedule
This screen administers a fixed schedule or a schedule that can change up to six times a day for each day of the week. This screen determines which CBC Service Trunk Group Allocation Plan will be in use at any given time. Available only if the ISDN trunk group Service Type is cbc and the ISDN trunk group Usage Alloc are enabled. Related links Service Type on page 512 Usage Alloc on page 516
Act Time
The time the usage allocation plan will become effective.
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Valid Entry
00:00 through 23:59
CBC Trunk Group Usage Allocation
Usage The time using the 24hour clock (military time). There must be at least one entry per day.
Allocation Plan Number
Valid Entry 1 to 3 blank
Usage
The CBC Trunk Allocation Plan that is in effect if a fixed usage method has been selected. Required if the allocation plan is fixed.
Fixed
Indicates whether the allocation plan will be fixed.
Plan #
Valid Entry 1 to 3 blank
Usage
The number of the usage allocation plan that will be in effect from the activation time until the activation time of the next scheduled plan change.
Scheduled
Indicates whether or not the allocation plans are implemented according to the administered schedule found on this page. If enabled, then there must be at least one entry in the schedule.
CBC Trunk Group Usage Allocation
This screen sets a minimum and maximum number of members for up to ten different Services/ Features for up to three different Usage Allocation Plans (1 to 3). Available only if the Service Type is cbc and Usage Alloc is enabled.
Related links Service Type on page 512 Usage Alloc on page 516
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ISDN Trunk Group
Max# Chan
Valid Entry 0 to 99 blank
Usage
The maximum number of members of an ISDN trunk group with a Service Type of cbc that a particular service or feature can use at any given time. This field must be administered if a service or feature has been entered in the Incoming Call Handling Treatment Table screen.
Min# Chan
Valid Entry 0 to 99 blank
Usage
The minimum number of members of an ISDN trunk group with a Service Type of cbc that a particular service or feature can use at any given time. The sum of the minimum number of members for all services or features must not exceed the total number of members of the trunk group.
Service/Feature
The ISDN services or features that can be requested at call setup time when using this trunk group. In addition to predefined services or features that can be received on a call by call basis, user-defined service types can be used.
Valid Entry 0 1 2 3 other
Usage Feature Service Incoming Outgoing Any services or features that are not explicitly specified
Wideband Support Options
Note: All B-channels that comprise the wideband call must reside on the same ISDN-PRI facility. Also, all trunk members in an ISDN trunk group with Wideband Support enabled must be from a TN464C (or later) circuit pack.
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Wideband Support Options
Contiguous
Available only if "N by DS-zero" (NXDS0) multi-rate service is enabled.
Valid Entry y
n
Usage
Specifies the "floating" scheme. NXDS0 calls are placed on a contiguous group of Bchannels large enough to satisfy the requested bandwidth without constraint on the starting channel (no fixed starting point trunk). Unavailable with H0 ISDN information transfer rate.
Specifies the "flexible" scheme. NXDS0 calls are placed on any set of B-channels on the same facility as long as the requested bandwidth is satisfied. There are no constraints, such as contiguity of B-channels or fixed starting points
Related links NxDS0 on page 545 NXDS0 on page 665
H0
If enabled, specifies the ISDN information transfer rate for 384-kbps of data that is comprised of six B-channels. When a trunk group is administered to support H0, the trunk or hunt algorithm to satisfy a call requiring 384-kbps of bandwidth uses a fixed allocation scheme.
H11
If enabled, specifies the ISDN information transfer rate for 1536-kbps of data that is comprised of 24 B-channels. When a trunk group is administered to support H11, the trunk or hunt algorithm to satisfy a call requiring 1536-kbps bandwidth uses a fixed allocation scheme.
H12
If enabled, specifies the ISDN information transfer rate for 1920-kbps of data that is comprised of 30 B-channels. When a trunk group is administered to support H12, the trunk or /hunt algorithm to satisfy a call requiring 1920-kbps bandwidth uses a fixed allocation scheme.
NxDS0
Enables or disables the "N by DS-zero" multi-rate service.
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ISDN Trunk Group
ISDN Trunk Group: Group Member Assignments
Note: When supporting DCS, Member Number Assignments must be the same between nodes. For example, Member #1 must be Member #1 at the far-end trunk group.
Administered Members (min/max)
The minimum and maximum member numbers that have been administered for this trunk group.
Code
The type of circuit pack physically installed or logically administered at the location to which this member is assigned. If no circuit pack is installed or administered at the port address, the field is blank.
Name
The name of the trunk group member. The name should identify the trunk unambiguously. Accepts up to 10 characters.
Example · The telephone number assigned to incoming trunks · The Trunk Circuit Identification number assigned by the service provider
Night
The night service destination for this trunk group member if different from the night service destination administered for the trunk group. Incoming calls are routed to this destination when the system is placed in night service mode.
Valid Entry A valid extension attd blank
Usage The extension of the night destination for the trunk. Calls go to the attendant when night service is active. Not administered
Related links Night Service on page 911
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ISDN Trunk Group: Group Member Assignments
Port
When using ISDN-BRI interfaces, B-channel 1 is the port number while B channel 2 is the port number plus 16. For example, if B channel 1's port number is 01A1002, then B channel 2's port number is 01A1018.
When using ISDN-PRI interfaces, the port number will be the one allied with the B-channel. For example, if the DS1 is located in 01A10, then B channel 1 will be 01A1001, B channel 2 will be 01A1002 and so forth.
Note:
When administering analog trunks connected to a TIM518, physical ports 17 to 24 are administered as ports 9 to 16 in Communication Manager.
Sfx
The model suffix for the type of circuit pack physically installed at the location to which this member is assigned. If no circuit pack is installed at the port address, the field is blank.
Sig Grp
Valid Entry digits
Usage
The signaling group of this trunk group member. Required for IP group members. If you administer a port that resides on a DS1 board and that DS1 board belongs to one and only one signaling group, you can leave the Signaling Group blank. The appropriate default signaling group number is inserted by Communication Manager. If a DS1 board is assigned to more than one signaling group, then you must enter a signaling group number. A trunk group can contain members from different signaling groups. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Related links Group Type on page 508 Signaling Group on page 514
Total Administered Members
The total number of members administered in the trunk group.
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Chapter 76: ISDN-BRI Trunk Circuit Pack
Administers an ISDN-BRI circuit pack. Example command: change bri-trunk-board n, where n is the board location.
ISDN-BRI Trunk Circuit Pack: page 1, TN2185 circuit pack
Cntry/Peer Protocol
The ISDN protocol standard that is applied.
Valid Entry 1 to 25
etsi QSIG blank
Usage
An entry of 10, 12, or 13 is equivalent to a Protocol Version of b administered for the DS1 Circuit Pack. All other administered values are equivalent to a Protocol Version of a administered for the DS1 Circuit Pack.
Equivalent to a Protocol Version of b administered for the DS1 Circuit Pack.
Required for a peer-slave or peer-master interface. Valid only when the Interface field is peer-slave.
Cannot be blank if an interface is administered.
Related links Protocol Version on page 232 Interface on page 549
DCP/Analog Bearer Capability
Valid Entry 3.1kHz speech
Usage
Indicates how to encode the Bearer Capability IE for an outgoing call originated by a DCP or analog endpoint.
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ISDN-BRI Trunk Circuit Pack: page 1, TN2185 circuit pack
Detect Slips
The system displays the field only if you set the Termination Type field to TE. Use the field to determine the state of health due to slips. This field is applicable only to the TN2185 boards.
ETSI CCBS
Available only if TSC SS Protocol is c.
Valid Entry none inco(ming) outg(oing) both directions
Usage Interface supports neither incoming nor outgoing ETSI CCBS. This is the default. Interface supports only incoming ETSI CCBS. Interface supports only outgoing ETSI CCBS. Interface supports incoming and outgoing ETSI CCBS.
Note:
When upgrading from a version of Communication Manager that is earlier than 5.1, this is the default.
Related links Group Type on page 508
Interface
Valid Entry network user peer-master peer-slave
Usage
Indicates whether a particular port is connected to a user, network or a peer interface. These entries are valid for the TN2185. Peer-slave is available only if the QSIG Basic Call Setup feature is enabled.
Interface Companding
The companding algorithm expected by the system at the far end.
Valid Entry a-law mu-law
Usage Algorithm expected at the far-end for E1 service. Algorithm expected at the far-end for T1 service.
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ISDN-BRI Trunk Circuit Pack
Layer 1 Stable
The system displays the field only if you set the Termination Type field to TE.
Valid Entry y n
Usage
The far-end network is stable at Layer 1.
The far-end network can drop Layer 1 after a call is completed and near-end ignores the Layer 1 disconnect message.
Location
Displays the TN2185 circuit pack location (PPCSS).
Name
The name used to identify the circuit pack. Accepts up to 15 alphanumeric characters.
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Port
The port number to which administered parameters apply.
Side
Valid Entry a b
Usage Determines how glare conditions are handled for a peer-slave interface.
Related links Interface on page 228
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ISDN-BRI Trunk Circuit Pack: page 1, TN2185 circuit pack
Synch Source
Valid Entry y
n
Usage
You can administer the board in the change synchronization screen as a primary or secondary Synch Source.
You cannot administer the board in the change synchronization screen as a primary or secondary Synch Source.
Note: The Synch Source field, when displayed, applies to TN2185, MM720, and MM721 boards.
T3 Timer Length (sec)
Valid Entry 1 to 127
Usage
Tells the TE side how many seconds to wait for an inactive Layer 1 to become active.
TEI
Valid Entry auto 0
Usage TEI is assigned automatically by the network. TEI is fixed.
Termination Type
When a MM720 media module is used as a trunk interface, and the MM720 supports both Line side and Trunk side of BRI, indicates whether the media module is to operate in Terminal or Network termination mode.
When you change the Termination Type of the MM720/MM721 media module from TE to NT, the system checks if the media module is administered as a clock synchronization source in the Synchronization Source screen, which is the circuit packet location. After the system completes the checks, the system displays the following error message if the board is administered as a synchronization source: This board is used for synchronization; cannot change Termination Type.
If you change the Termination Type of the MM720/MM721 media module from NT to TE and set any of the interface fields to network or peer-master for a port, the system displays the following
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ISDN-BRI Trunk Circuit Pack
warning message: TE Termination Type not recommended with network Interface types.
Likewise, if you change the Termination Type of the MM720/MM721 media module from TE to NT and set any of the interface fields to user or peer-slave for a port, the system displays the following warning message: NT Termination Type not recommended with network Interface types.
However, you can still submit these changes, if required.
Valid Entry TE
NT
Usage
Terminal Endpoint termination. The MM720 provides the TE side of the BRI interface. This is default.
Network Termination. The MM720 provides the NT side of the BRI interface. The NT type board cannot be used as a clock synchronization source.
ISDN-BRI Trunk Circuit Pack: page 1, TN556B or TN2198 circuit packs
The following field descriptions are unique to the ISDN-BRI Circuit Pack screen with a TN556B or TN2198 circuit pack. The following fields do not display with a TN556B or TN2198 circuit pack:
· T3 Timer Length (sec) · Synch Source · Layer 1 Stable · Detect Slips
DCP/Analog Bearer Capability
Valid Entry 3.1kHz speech
Usage
Indicates how to encode the Bearer Capability IE for an outgoing call originated by a DCP or analog endpoint.
Cntry/Peer Protocol
The ISDN protocol standard that is applied.
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ISDN-BRI Trunk Circuit Pack: page 2
Valid Entry 1 to 25
etsi QSIG blank
Usage
An entry of 10, 12, or 13 is equivalent to a Protocol Version of b administered for the DS1 Circuit Pack. All other administered values are equivalent to a Protocol Version of a administered for the DS1 Circuit Pack.
Equivalent to a Protocol Version of b administered for the DS1 Circuit Pack.
Required for a peer-master interface.
This field cannot be blank if an interface is administered.
Related links Interface on page 553 Protocol Version on page 232
Interface
Indicates whether a particular port is connected to a user/network or a peer interface. These entries are valid for the TN556B.
Valid Entry network peer-master
Usage User/network connection. Peer interface connection. Available only if QSIG Basic Call Setup is enabled
Side
Valid Entry a b
Usage Determines how glare conditions are handled for a peer-slave interface.
Related links Interface on page 228
ISDN-BRI Trunk Circuit Pack: page 2
Note: If administering a TN2185 circuit pack, the system displays 8 ports; otherwise, the system displays 12 ports.
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ISDN-BRI Trunk Circuit Pack
Note: You cannot change the Endpt Init, SPID, or Endpt ID port parameters unless that port is busied out or unadministered. It is possible to change all other fields on this page even if the corresponding port is active. If the Interface field on page 1 contains a valid value when the screen is submitted, the contents of the fields on page 2 for that port are validated. If the Interface field is blank when the screen is submitted, the fields on this page for that port reset to their default values.
Directory Number
The directory numbers assigned to the interface and allocated to two separate endpoints. This field must be administered in pairs. Accepts up to 10 characters.
Endpt ID
Valid Entry 00 to 62
Usage
The Endpoint Identifier expected by the far end. Communication Manager prevents changing this field unless the port is busied out or unadministered. Leading zeroes are significant and must not be ignored.
Endpt Init
Indicates whether the far end supports endpoint initialization. Communication Manager blocks you from changing this field unless the port is busied out or unadministered.
Valid Entry y n
Usage Requires that an SPID be administered. Requires that an SPID and Endpt ID not be administered.
Related links Endpt ID on page 554 SPID on page 555
Interworking Message
Determines what message Communication Manager sends when an incoming ISDN trunk call is routed over a non-ISDN trunk group.
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Valid Entry PROGress
ALERTing
ISDN-BRI Trunk Circuit Pack: page 2
Usage Requests the public network to cut through the B-channel and let the caller hear tones such as ringback or busy tone provided over the non-ISDN trunk. Normallyselected value. Causes the public network in many countries to play ringback tone to the caller. This value is used only if the DS1 is connected to the public network, and it is determined that callers hear silence (rather than ringback or busy tone) when a call incoming over the DS1 interworks to a non-ISDN trunk.
Max NCA TSC
Valid Entry 0 to 63
Usage
The maximum number of Non-Call-Associated Temporary Signaling Connections allowed on this BRI D-channel.
Port
The port number to which administered parameters apply.
SPID
The Service Profile Identifier (SPID) expected by the far end. Accepts up to 12 characters. Communication Manager prevents changing this field unless the port is busied out or unadministered. The only protocol supported for SPID initialization is Country Code 1. Trunks are not put in service if SPID installation is unsuccessful. Leading zeroes are significant and must not be ignored.
XID Test
If enabled, the far end supports the Layer 2 XID test.
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Chapter 77: Language translations
The Language Translations screens display messages and their translations. Telephones display pre-translated messages from Communication Manager, which are available in the following languages:
· English · French · Italian · Spanish You can assign translations in any other language by using the Language Translations screens, but this method is deprecated. Instead, use Avaya Message Editing Tool (AMET) to assign translations for user-defined telephone messages. If you use an editor other than AMET, the system does not update the Phone Message Files correctly and the files are not installed. For more details, see Avaya Message Editing Tool (AMET) Job Aid. Note: If you type user-defined in the Display Language field on the Station screen or the Attendant Console screen and messages are not defined for these screens, the system displays a string of asterisks on all display messages.
Warning: If you use AMET, Communication Manager blocks access to the SAT user-defined translationsrelated screens. Therefore, do not use the Language Translations screens.
Warning: If you do not use AMET, do not edit the user-defined translations-related screens through SAT. You can download AMET from http://www.avaya.com.
Automatic Wakeup
Use this field to replace the English text for Automatic Wakeup messages that the system displays on the telephone screen. This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters.
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Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Button labels
Button labels
Use this field to replace the English text for Button labels that the system displays on the telephone screen.
This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters.
Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Enhanced Abbreviated Dialing
Use this field to administer the Enhanced Abbreviated Dialing messages that the system displays on the telephone screen.
This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters.
Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
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Language translations
Leave Word Calling
Use this field to replace the English text for Leave Word Calling messages that the system displays on the telephone screen. This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters. Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Malicious Call Trace
Use this field to replace the English text for Malicious Call Trace messages that the system displays on the telephone screen. This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters. Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Miscellaneous call identifiers
Use this field to replace the English text for miscellaneous call identifiers that the system displays on the telephone screen. This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters. Related links Message 12 on page 562 Message 15, 16, 18 on page 562
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Message 4 on page 561 Message 6 on page 562
Miscellaneous features
Miscellaneous features
Use this field to replace the English text for miscellaneous features that the system displays on the telephone screen.
This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters.
Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Property Management Interface
Use this field to replace the English text for property management interface messages that the system displays on the telephone screen.
This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters.
Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Self Administration
Use this field to replace the English text for self administered messages that the system displays on the telephone screen.
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Language translations
This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters. Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Softkey Labels
Use this field to replace the English text for softkey labels that the system displays on the telephone screen. This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters. To provide unique labels for abbreviated dialing button types for softkey labels, Communication Manager replaces the last two characters with digits for the 12-key 8400 and 15-key 8434D telephones. On this screen, the digits following the AD are derived from the button position. If the first button is an AD button, then it is AD1 and the 15th button is AD15. All the AD buttons between 1 and 15 have the position number appended to AD. Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Transfer Conference
Use this field to replace the English text for transfer conference messages that the system displays on the telephone screen. This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters. Related links Message 12 on page 562
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Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
View Buttons
View Buttons
Use this field to replace the English text for buttons that the system displays on the telephone screen.
This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters.
Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Vustats
Use this field to replace the English text for Vustats that the system displays on the telephone screen.
This screen contains an English version of the display, an explanation of the English message, and a translation of the English message. A long message can be shortened on telephones that display fewer than 32 characters.
Related links Message 12 on page 562 Message 15, 16, 18 on page 562 Message 4 on page 561 Message 6 on page 562
Message 4
The character "^" is a place holder.
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Language translations
English Text
^-party conference in progress
Replacement Info "^" is replaced with the number of parties currently on the conference call.
Manually change ~ to ^ in any user-defined language. The software does not update manually.
Message 6
The character "^" is a place holder.
English Text
Select line ^ to cancel or another line.
Replacement Info "^" is replaced with the letter of the line that is on soft hold.
Manually change ~ to ^ in any user-defined language. The software does not update manually.
Message 15, 16, 18
The character "^" is a place holder.
English Text
Select line ^ to add party.
Replacement Info "^" is replaced with the letter of the line that is on soft hold.
Message 12
The character "^" is a place holder.
English Text
Select line ^ to add party.
Replacement Info "^" is replaced with the letter of the line that is on soft hold.
Manually change ~ to ^ in any user-defined language. The software does not update manually.
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Chapter 78: Listed Directory Numbers
With Listed Directory Numbers, Direct Inward Dialing (DID) numbers can be treated as public Listed Directory Numbers (LDNs). When one of these numbers is direct inward dialed, the calling party is routed to the attendant. The attendant display indicates a Listed Directory Number call and the name associated with the dialed extension. Example command: change listed-directory-numbers
Ext
An extension number containing 16 digits.
Name
The name used to identify the Listed Directory Number. Accepts up to 27 alphanumeric characters.
Night Destination
A valid assigned extension number that receives calls to these numbers when Night Service is active. Accepts up to sixteen digits.
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
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Chapter 79: Locations
This screen provides daylight saving time displays. You can set the area code for each location and administer different location information for each location. If the Multiple Locations feature is enabled, you can administer up to 2000 location specifications, depending on the configuration of Communication Manager. Otherwise, information for Location 1 applies to all locations. You can use the following command to modify the fields on this screen: change locations
ARS FAC
The Feature Access Code (FAC) for accessing Automatic Route Selection (ARS). Any valid FAC format is acceptable, up to four digits. Characters * or # are permitted, but only in the first position. Many locations are expected to share the same access code.
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Related links Feature Access Code (FAC) on page 253
ARS Prefix 1 Required for 10-Digit NANP Calls?
ARS Prefix 1 Required for 10-Digit NANP Calls?
If enabled, a 1 must be dialed before all 10-digit NANP calls.
Attd FAC
The Feature Access Code (FAC) for connection to the attendant. Accepts up to two digits. Characters * or # are not permitted. Many locations are expected to share the same access code.
Available only if an Attendant Access Code has first been administered either for the Dial Plan or as a standard FAC.
Note:
Within a dial plan, FAC/DAC codes and extensions cannot both start with the same first digits. Either the FAC/DAC entries or the block of extensions must be changed to have a different first digit.
Related links Call Type on page 210 Attendant Access Code on page 255
Disp Parm
The administered display parameters for the location. Related links
Display Parameters on page 221
Loc Parm
Required when administering multiple locations.
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Locations
Valid Entry 1 to 50 blank
Usage
The number of the corresponding Location Parameter set for this location. Default is blank.
Loc Number
Valid Entry 1 to 2000
Usage The location number.
Name
A name you use for the location. Names are easier to remember than location numbers. Accepts up to 15 alphanumeric characters.
City/Area
The three-digit numbering plan area code for each location.
Prefix
Valid Entry 0 to 99999
Usage
Location prefix used with the Uniform Dialing Plan to identify the location of an originating call. All or part of the prefix is added to the front of the dialed string.
Related links Insert Digits on page 984
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Proxy Sel Rte Pat
Valid Entry 1 to 999 blank
Usage The routing pattern used to get to the proxy server.
Related links Route Pattern on page 683
Proxy Sel Rte Pat
DST
Required for each administered location.
Valid Entry 0 1 to 15 blank
Usage No Daylight Saving. The number for the Daylight-Saving Rule applied to this location.
Timezone Offset
Specifies how much time each location differs from the system time. Required for each administered location. Use +00:00 for the time zone offset for a single location media server.
Valid Entry + -
Usage The time set on this location is a certain amount of time ahead of the system time. The time set on this location is a certain amount of time behind the system time.
Valid Entry 0 to 23
Usage The number of hours difference between this location and system time.
Valid Entry 0 to 59
Usage The number of minutes difference between this location and system time.
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Chapter 80: Location Parameters
Sets or changes certain administrable characteristics that determine part of a location's behavior. These include recall timing intervals and loss plans for two-party and conference calls. Multiple instances of the Location Parameters screen are accessible if Multiple Locations is enabled for the system. If Multinational Locations is enabled, Location Parameters 2-50 contain the same fields as for Location Parameters 1. If Multinational Locations is disabled, the system does not display the following fields for Location Parameters 1:
· Tone Generation Plan · DCP Terminal-parameters Plan · Country Code for CDR
Location Parameters: page 1
Analog Ringing Cadence
The country code identifies the ringing cadence to be used by analog telephones in the system.
Valid entry 1 to 25
Usage The location country code.
Note:
This field must be set to 1 (US) for the Message Waiting Indicator on the Station to be set to neon.
Analog Line Transmission
The country code identifies the transmission and signaling parameters.
Valid entry 1 to 25
Usage The location country code.
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Location Parameters: page 1
Companding Mode
The companding algorithm used by system hardware.
Valid Entry A-Law Mu-law
Usage Generally used outside the U.S. Generally used in the U.S.
Convert ISDN Public Numbers to International for Display
If you set the field to values other than no, Communication Manager expands the number to the international numbering format. The default value of this field is no.
Valid entries All-stations All-digital DCP stations H.323 stations No
Usage
Communication Manager expands calling and connected ISDN numbers to international format when the number is displayed on any station.
Communication Manager expands calling and connected ISDN numbers to international format when the number is displayed on any digital station.
Communication Manager expands calling and connected ISDN numbers to international format when the number is displayed on any DCP station.
Communication Manager expands calling and connected ISDN numbers to international format when the number is displayed on any H.323 station.
Calling and connected ISDN numbers appear unchanged on stations with displays.
Country code for CDR
Available only if the Multinational Locations feature is enabled for the system.
Valid Entry 1 to 999
Usage
The country code to be used for Call Detail Recording information for a location. Default is 1.
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Location Parameters
Date Format on Terminals
The Date Format on Terminals field enables you to set the date format on the terminals, based on its location. The system displays this field only if you set the Multinational location field on the System Parameter Customer Options screen to y.
Valid Entry mm/dd/yy dd/mm/yy yy/mm/dd
Usage (month/day/year) (day/month/year) (Year/month/day)
DCP Terminal-parameters Plan
This field corresponds to the DCP terminal transmission parameters administered for the location. The system displays this field only if the Multinational Locations feature is enabled.
Valid entry 1 to 25
Usage The terminal-parameters plan number. The default value is 1.
End OCM After Answer (msec)
If the End OCM After Answer field contains a non-blank value, Communication Manager starts the End OCM timer when Communication Manager receives an answer signal. The End OCM timer ensures that an outgoing call using OCM call classification is answered by an agent or an announcement within a specified time.
Valid Entry 100 to 25,000
blank
Usage
The timeout value is in milliseconds. If the timer expires, Communication Manager disconnects the call classifier and connects the call to the administered intercept extension. If the call classifier classifies the call before the timer expires, the timer is cancelled and the call routed or treated appropriately.
Indicates that the timer has no limit.
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Location Parameters: page 1
End of OCM Intercept Extension
Valid Entry extension
blank
Usage
The extension number Communication Manager connects the call to when the End OCM After Answer field expires. The extension number can be a recorded announcement, a vector directory number, or a hunt group extension.
Indicates that the extension number is empty. If the End OCM After Answer field is set to a non-blank value, the End of OCM Intercept Extension field cannot be left blank.
International Access Code
An up to five-digit International Access Code. Default is blank.
Local E.164 Country Code
An up to three-digit E.164 Country Code. Default is blank.
For a list of country codes, see the International Telecommunications Union List of ITU-T Recommendation E.164 Assigned Country Codes.
Long Distance Access Code
Valid Entry 0 to 9
blank
Usage
The long distance access code you want the system to prefix to the telephone number. Accepts up to five digits.
Not administered. This is the default.
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Location Parameters
Off-PBX Feature Name Extension Set
Valid Entry
0 to 10 for a medium configuration
1 to 99 for a large configuration
blank
Usage Feature Name Extension (FNE) set that should be used for location based FNEs.
Default FNE set is used. This is the default.
Time Format on Terminals
With the Time Format on Terminals field, you can control the format for displaying the time on non-SIP endpoints. The system displays the Time Format on Terminals field even when the Multinational Locations feature is not enabled. Therefore, if all users are in a single country, you can change the time format for all users. To use different time formats for users in different locations, you must enable the Multinational Locations feature.
You can change the time format for SIP endpoints from the endpoint itself.
Valid entry 12-hour
24-hour
based-on-language
Usage To display the time in the 12-hour format on non-SIP display sets.
With this option, the system overrides the display language and SA8885 settings. To display the time in the 24-hour format on non-SIP display sets.
With this option, the system overrides the display language and SA8885 settings. To use the display language and SA8885 to determine the time format for non-SIP display sets.
This option is the default value.
Tone Generation Plan
The value in this field corresponds to the tone generation characteristics administered for the location. The system displays this field only if the Multinational Locations feature is enabled for the system.
Valid Entry 1 to 25
Usage The tone-generation plan number. The default value is 1.
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Location Parameters: page 1
RECALL TIMING
Disconnect Timing (msec)
Available only if a Flashhook Interval is not required.
Valid Entry
80 to 1250 (in increments of 10)
Usage
An on-hook that lasts for a period of time less than this value will be ignored; greater than or equal to this value is regarded as a disconnect.
Flashhook Interval
Enables or disables requiring a flashhook interval (recall window).
Forward Disconnect Timer (msec)
Valid Entry
25 to 1500 (in increments of 25)
Usage
Specifies the duration of a momentary disconnect sent by the server/switch to an analog station user when that user is the last party still off-hook on a call.
Lower Bound (msec)
Available only if a flashhook interval is required.
Valid Entry
80 to 1250 (in increments of 10)
Usage
The lower bound of the station-to-switch recall signal timing interval in milliseconds. Specifies the lower bound of the flashhook interval.
MF Interdigit Timer (sec)
Valid Entry 1 to 255
Usage
The maximum number of seconds Communication Manager waits for the first forward signal (digit) to arrive, and for subsequent digits to arrive. Intercept returns to the calling party if this timer expires. This number must be less than the number of seconds administered for the short interdigit timer. Applies only to multifrequency signaling trunks.
Outgoing Shuttle Exchange Cycle Timer (sec)
Available only if the Incoming Call Type is group-ii-mfc or non-group-ii-mfc and the Outgoing Call Type is group-ii-mfc or none for the Multifrequency Signaling-Related System Parameters. This field applies only to multifrequency signaling calls made from Avaya Communication Manager.
Valid Entry 1 to 25
Usage
The number of seconds to time an exchange cycle (starts when the far end requests a digit until Avaya Communication Manager sends the requested digit.
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Location Parameters
Upper Bound (msec)
Available only if the flashhook interval is required.
Valid Entry
80 to 1250 (in increments of 10)
Usage
The upper bound of the flashhook interval. Specifies the upper bound of the stationto-switch recall signal timing interval in milliseconds. A flash of 50 msec to 130 msec is always acceptable from a 2500-type set regardless of the setting of the Upper and Lower bounds and is treated as the digit one.
Replace International Access Code with '+'
If you set the field to values other than no, Communication Manager the plus sign (+) instead of the configured international access code in front of international calling and connected numbers received from an ISDN trunk group. The default value of this field is no.
Valid entries All-stations
All-digital
DCP stations H.323 stations No
Usage
Communication Manager adds the plus sign (+) to international calling and connected ISDN numbers that appear on all stations that includes analog with a display.
Communication Manager adds the plus sign (+) to international calling and connected ISDN numbers that appear on digital stations with a display.
Communication Manager adds the plus sign (+) to international calling and connected ISDN numbers that appear on DCP stations.
Communication Manager adds a plus sign (+) to international calling and connected ISDN numbers that appear on the H.323 stations.
International calling and connected ISDN numbers appear with the international access code, and no plus sign (+) on stations with a display.
Location Parameters: page 2
LOSS PLANS
2-Party Loss Plan/Tone Loss Plan
This field provides the default values for digital loss plan and for n-party conference loss.
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Location Parameters: page 3
Valid entry 1 to 25
Usage A country code for the location.
Customize
Enables customization on the corresponding loss plan table. When Customize is enabled for the End-to-End total loss (dB) in a n-party conference, values can be changed by the administrator. When Customize is disabled, the End-to-End total loss (dB) in a n-party conference values are reset to the values that they would have had under the 2 Party Loss Plan. Available only if Digital Loss Plan Modification is enabled for the system.
End-to-End total loss (dB) in a n-party conference
Valid Entry 0 to 99
Usage
Provides total loss for a conference call with the designated number of parties.
The higher the number listed for a call with a fixed number of parties, the more loss Communication Manager adds into a conference call with that number of parties; therefore, the conference call is quieter.
Note:
The End-to-End total loss for multi-party conference calls that is administered for this field is not always applied to a specific call.
Inter-location Loss Group
When inserting loss for a call, the server treats parties on the call who are in separate locations as if the location with the most parties were connected by an equal number of IP tie trunks as there are parties at other locations. The field specifies the digital loss group number that is used by these virtual IP tie trunks. The system displays this field only if the Multinational Locations feature is enabled.
Valid entry 1 to 19
Usage
The digital loss group number to use on inter-location calls involving this location. The default value is 18.
Location Parameters: page 3
FROM / TO
Valid Entry 3 to 15
Usage
Identifies the variable digital loss values. An unsigned number is a decibel loss, while a number preceded with a minus sign is a decibel gain.
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Chapter 81: Login Administration
Beginning with Communication Manager 4.0, there is no longer a Login Administration screen. For details on screens used for login administration, see Maintenance Commands for Avaya Aura® Communication Manager, branch Gateways and Servers and "AAA Services" in Avaya Aura® Communication Manager Feature Description and Implementation.
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Chapter 82: Logging levels
Administers logging of SAT activities. Specifies that commands associated with specific actions shown on this screen are logged by the system. The amount of detail to be logged is the same for all enabled actions and is specified by the Log Data Values field.
Note: The defaults on this screen provide the same amount and type of logging as in Communication Manager releases prior to 4.0. Example command: change logging-levels Related links Log Data Values on page 577
Logging Levels: page 1
Enable Command Logging
If enabled, SAT activity is logged for selected commands.
Log Data Values
Valid Entry none new both
Usage Only the object, the qualifier, and the command action are logged. The new value of any field is logged. The old value is not logged. Both the prior field value and the field value after the change are logged.
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Logging levels
Logging Levels: page 2
Log All Submission Failures
Form submission failures due to a security violation are always logged and are unaffected by this field.
Valid Entry y
n
Usage
Record submission failures on the history log. An event is logged when Avaya Communication Manager rejects a form submission for any reason, such as an invalid entry in a field or a missing value.
Submission failures are not recorded on the history log.
Log CTA/PSA/TTI Transactions
Enables or disables logging transactions when extensions and physical telephones move between ports without additional administration from the administrator of Communication Manager. If enabled, the system records Customer Telephone Activation (CTA), Personal Station Activation (PSA), and TTI transactions in the system history log. Available only if Terminal Translation Initialization (TTI) is enabled for the system. Related links Terminal Trans. Init. (TTI) on page 841
Log IP Registrations and Events
If enabled, the IP registrations can be logged in the history log.
Log PMS/AD Transactions
If enabled, the system records Property Management System (PMS) and Abbreviated Dialing (AD) events to the log.
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Chapter 83: Loudspeaker Paging
The Loudspeaker Paging screen administers voice paging, deluxe voice paging, and chime paging. Note: To set up paging on a branch gateway, connect the paging system to a port on an MM711 and administer the port as an analog station on the Station screen. No entries on the Loudspeaker Paging screen are required.
Example command: change paging loudspeaker
CDR
If enabled, Communication Manager collects CDR data on the paging ports.
Code Calling -- COR
Valid Entry 0 to 995 blank
Usage Assigns a Class of Restriction to a paging zone for chime paging. An unused paging zone.
Code Calling Playing Cycles
Valid Entry 1 to 3 blank
Usage The number of times a chime code plays when a user makes a chime page. Cannot be blank when administering chime paging (code calling).
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Loudspeaker Paging
Code Calling -- TAC
Assigns a Trunk Access Code (TAC) to a paging zone for chime paging. Users dial this code to make a page to a specific zone. One TAC must be assigned to each zone used. Two zones cannot have the same TAC. A TAC in the zone designated ALL means that users can activate speakers in all the zones by dialing that code.
Valid Entry 1 to 4 digits * # blank
Usage A Trunk Access Code (TAC) allowed by the dial plan. Can be used as first digit. Can be used as first digit. An unused paging zone.
Code Calling -- TN
Valid Entry 1 to 100
Usage
Assigns a paging zone to a tenant partition for chime paging when Tenant Partitioning is in use.
Related links Tenant Partitioning on page 840
Location
Assigns a descriptive name for the physical location corresponding to each zone when Tenant Partitioning is in use. Accepts up to 27 characters. Example Typical entries might be "conference room A", "warehouse", or "storeroom".
Port
Assigns a port on an auxiliary trunk circuit pack to a paging zone.
Valid Entry 1 to 64
Usage First and second characters are the cabinet number
Table continues...
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Valid Entry A to E 0 to 20 0104 (Analog TIE trunks)
0131 1 to 250 VI to V9 01 to 31 blank
Usage Third character is the carrier Fourth and fifth character are the slot number Six and seventh characters are the circuit number
Gateway Module Circuit An unused paging zone
Voice Paging -- COR
Voice Paging -- COR
Valid Entry 0 to 995 blank
Usage Assigns a Class of Restriction to a paging zone for voice paging. An unused paging zone.
Voice Paging -- TAC
Use this field to assign a Trunk Access Code (TAC) to a paging zone for voice paging. Users dial this code to make a page to a specific zone. One TAC must be assigned to each zone. Two zones cannot have the same TAC. If you assign a TAC to the zone designated ALL, users can activate speakers in all the zones by dialing the code.
Valid entry digits * # blank
Usage Up to four digit TAC according to the dial plan Can only be used as first digit Can only be used as first digit An unused paging zone
Voice Paging Timeout (sec)
Limits the duration of voice pages. When this interval ends, calls are disconnected. To determine the best setting, time the typical pages that are expected to broadcast and then add another 4 to 5 seconds.
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Loudspeaker Paging
Valid Entry 10 to 60 seconds blank
Usage The maximum number of seconds any page lasts. The field cannot be blank when voice paging is administered.
Voice Paging -- TN
Valid Entry 1 to 100
Usage
Assigns a paging zone to a tenant partition for voice paging if Tenant Partitioning is in use.
Related links Tenant Partitioning on page 840
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Chapter 84: Maintenance-related system parameters
This screen is described in Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and Servers.
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Chapter 85: Media-Gateway
This screen is described in Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and Servers.
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Chapter 86: Media Server
Use the Media Server screen to add media servers to Communication Manager. To modify the fields on the Media Server screen, run the following command: change media-server n, where n the media server number. Where, n has a value from 1 to 250. To view the status of the Media Server screen, run the following command: status media-server .
Signaling Group
The SIP signaling group specifies the SIP signaling group that Communication Manager uses to communicate with the media server. The signaling-group must be populated with a Peer Server of type AMS. The signaling-group number entered on the media-server form may not be assigned to any other media-server.
Valid entry Node Name Network Region Location Announcement Storage Area
Usage
Far-End Node Name of the signaling group form.
Far-End Network Region from the signaling group form.
Location obtained from the Network Region form.
A display-only field that specifies the location on the mediaserver where the announcements for Communication Manager are stored. This value is same for all media-servers used by a specific Communication Manager. If the media-server is shared between different Communication Managers, each Communication Manager has a separate announcement storage area.
Voip Channel License Limit
The Voip Channel License Limit field represents the maximum number of simultaneous Voip channel licenses, which can be used on the specific media server. The field represents an upper limit. You can type a value from 1 to 9999 or leave the field blank.
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Media Server If the field is left blank, the upper limit of simultaneous Voip channel licenses that can be used on the media-server are determined as follows: · By the capacity of the media-server · By the availability of Voip channel licenses
Dedicated Voip Channel Licenses
The Dedicated Voip Channel Licenses field represents the number of Voip channel licenses dedicated to the specified media server. Dedicated Voip Channel Licenses field is left blank or has a value between 1 and 9999. Blank implies no channel licenses are dedicated to the mediaserver.
Capacity Profile Max Sessions
Capacity profile max session is a value provided by AMS as part of its capacity profile and is given to the communication manager when the AMS comes into service. This value represents the maximum number of sessions of any type (audio, video, control), which this AMS server could establish. There are no limitations on the number of sessions, but rather a hard limit assuming an optimal selection of codecs and enough processing power and memory.
Capacity Profile MPU Estimate
Capacity profile Media Processing Unit (MPU) estimate is a value provided by AMS as part of its capacity profile. It is given to Communication Manager when the AMS comes into service. This is an estimate of the MPU capacity of the physical server. Capacity profile MPU value is estimated by AMS based on server type, clock speed, number of physical processor cores, logical processors, RAM, disc space, and NIC speed. Different AMS actions require different amount of MPU. For example, a simple G711, non-encrypted channel requires 1 MPU. More complex audio or video codecs require higher MPUs.
Cluster Type
AMS is configured as 1 of 3 available cluster types: Stand-alone, High-available, or Load-sharing. When AMS is a client of the Communication Manager, only the Stand-alone or High-available cluster types are supported. If AMS is incorrectly configured as part of a Load-sharing cluster, Communication Manager will not place the AMS into service. In this case, the state field of the status form indicates this with the new string: "OOS-ILLEGAL loadshare cluster". Additionally,
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Cluster Type the "list media-server" report will also indicate the AMS is out-of-service with this new state: "OOSLS".
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Chapter 87: Message Waiting Indication Subscriber Number Prefixes
If the voice mail server is an AUDIX server connected using BX.25, you must fill in the Message Waiting Indication (MWI) Subscriber Number Prefixes to send or cancel a Message Waiting Indication to a served user. The served user is a user on the Private Telecommunication Network (PTN) to whom the MWI is sent on activation by the Message Center or Message Center PBX.
Using the QSIG MWI Supplementary Service, Message Center/Message Center PBX sends MWI Activate or MWI Deactivate message to the served user as a result of storage or retrieval of messages. Also, the served user is allowed to query the Message Center for MWI status.
Example command: change isdn mwi-prefix
AUDIX Mach ID
Valid entry 1 to 99
blank
Usage
AUDIX number serving the switch of the AUDIX number field from the Switch Interface Administration screen.
Available only if all the other records on the screen are blank.
Inserted Digits
Accepts up to 12 digits.
Valid entry 0 to 9
blank
Usage
Digits prefixed to the AUDIX subscriber number to provide complete subscriber number to the Message Center PBX.
No digits prefixed to the AUDIX subscriber number.
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Machine ID
Machine ID
Available only if Machine ID is assigned to the qsig-mwi application in the Processor Channels Communication Interface.
Valid entry 1 to 20
Usage Unique ID associated with each processor channel qsig-mwi application
Message Center ID
Accepts up to 20 digits. These are the Message Center ID digits to be sent a QSIG MWIActivate or MWIDeactivate message. It is necessary to enter the Message Center ID only if the system at the far-end of the QSIG connection needs it.
Routing Digits
Accepts up to four digits.
Valid entry 0 to 9, *, # blank
Usage
Digits comprising the AAR or ARS access code.
No Routing Digits prefixed to the AUDIX subscriber number. For example, if UDP is used to route the call.
Send QSIG Message Center ID
Valid Entry y
n
Insert a message center ID in the MWI Activate and MWI Deactivate messages. For SIP-connected messaging servers, Communication Manager can be administered to insert a Message Center ID over QSIG connections to other servers by entering the Message Center ID, along with the Local Node Number for this Communication Manager server, in any record of this form.
Does not insert the message center ID in the MWI Activate and MWI Deactivate messages.
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Chapter 88: Mode Code Related System Parameters
Establishes parameters associated with the Mode Code Voice Mail System Interface. Available only if the Mode Code Interface feature is enabled. Example command: change system-parameters mode-code Related links
Mode Code Interface on page 308
MODE CODES (FROM SWITCH TO VMS)
Direct Dial Access-Trunk
The mode code that the media server or switch sends when an external caller dials the Voice Mail System (VMS) access number. Accepts up to six digits. Also accepts #, *, and #00.
Direct Inside Access
The mode code that the media server or switch sends when a caller at an internal extension dials the Voice Mail System (VMS) access number. Accepts up to six digits. Also accepts #, *, and #00.
External Coverage
The mode code that the media server or switch sends when an external caller tries to reach a user at another extension and the call goes to the user's voice mail coverage. Accepts up to six digits. Also accepts #, *, and #00.
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OTHER RELATED PARAMETERS
Internal Coverage
The mode code that Communication Manager sends when an internal caller tries to reach a user at another extension and the call goes to the user's voice mail coverage. Accepts up to six digits. Also accepts #, *, and #00.
Refresh MW Lamp
The mode code that Communication Manager sends during a system level 3 or higher reset that requests the VMS to refresh the Message Waiting (MW) lamps. Accepts up to six digits. Also accepts #, *, and #00.
System In Day Service
This value is used by the Voice Mail System (VMS) to indicate that Communication Manager has changed from Night to Day Service. Accepts up to six digits. Also accepts #, *, and #11.
System In Night Service
This value is used by the Voice Mail System (VMS) to indicate that Communication Manager has changed from Day to Night Service. Accepts up to six digits. Also accepts #, *, and #12.
OTHER RELATED PARAMETERS
DTMF Duration On
Valid Entry
75 to 500 -- in multiples of 25
Usage
The duration in milliseconds of mode code digits sent to the VMS. This field cannot be blank.
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Mode Code Related System Parameters
Off
Valid Entry
Between 75 and 200 -- in multiples of 25
Usage
The pause in milliseconds between mode code digits as they are sent to the VMS. This field cannot be blank.
Remote VMS Extensions- First
The first remote UDP VMS hunt group extension. Available only if Mode Code for Centralized Voice Mail is enabled for the system.
Related links Mode Code for Centralized Voice Mail on page 837
Remote VMS Extensions - Second
The second remote UDP VMS hunt group extension. This extension cannot be the same as the first Remote VMS Extension. Available only if Mode Code for Centralized Voice Mail is enabled for the system.
Related links Mode Code for Centralized Voice Mail on page 837
Sending Delay
Valid Entry
75 to 1000 -- in multiples of 25
Usage
The delay in milliseconds between the time answer supervision is received from the VMS and the time the first mode code digit is sent. This field cannot be blank.
VMS Hunt Group Extension
The extension of a hunt group containing VMI extensions. A check is made to verify that a valid hunt group extension is entered, but a check is not made to verify that the hunt group members are VMI extensions.
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Chapter 89: Modem Pool Group
There are two types of conversion resources for Modem Pooling. The first type, an integrated conversion resource, is a circuit pack that emulates a Trunk Data Module connected to a 212A-type modem. Two conversion resources are on each circuit pack. The second type, a combined conversion resource, is a separate Trunk Data Module and modem administered as a unit. The Trunk Data Module component of the conversion resource can be either a Modular Trunk Data Module (MTDM) or 7400A Data Module and connects to a digital port using Digital Communications Protocol (DCP); the modem connects to an analog port.
Note: The Speed, Duplex, and Synchronization fields cannot be filled out for the "integrated" pooled modem screens but can be assigned on the "combined" pooled modem screen. The integrated conversion resource automatically will adjust its speed and synchronization to the endpoint it is connected to. In synchronous mode, the integrated modem pool can operate at 1200 baud. In asynchronous mode, it can operate at 300 or 1200 baud. Full-duplex operation is always used. Example command: change modem-pool n, where n is the modem pool number.
Answer Supervision Timeout (sec)
Available only with a combined type modem pool.
Valid Entry 1 to 255 0
Usage The number of seconds to wait before the far-end answers. No answer supervision
Related links Group Type on page 594
CF-CB Common
If enabled, the CF and CB leads on the conversion resource are logically connected. Available only with an integrated type modem pool.
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Modem Pool Group
Related links Group Type on page 594
Direction
The direction of the call for which the modem pool operates. Available only with a combined type modem pool.
Valid Entry incoming outgoing two-way
Usage Converts an analog signal to digital for the data endpoint Converts analog to digital (or digital to analog) for data calls Allows incoming and outgoing data communication
Related links Group Type on page 594
Duplex
The duplex mode of the conversion resources in the group. Entry required for a combined type modem pool.
Valid Entry full half
Usage Can talk and listen at the same time Cannot talk and listen at the same time
Related links Group Type on page 594
Group Number
The modem pool group number.
Group Type
The type of physical model pool.
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Valid Entry integrated combined
Hold Time (min)
Usage Maps to the Pooled Modem circuit pack Maps to an external modem pool when a data module and a modem exists
Hold Time (min)
Valid Entry 1 to 99
Usage
The maximum number of minutes that a conversion resource in the group can be held while a call waits in a queue or reserved after Data Call Preindication.
Loss of Carrier Disconnect
If enabled, permits conversion resource to disconnect if it detects a dropped carrier. Available only with an integrated type modem pool. Related links Group Type on page 594
Modem Name
The name of the modem pool. Accepts from one to six characters. Available only with a combined type modem pool.
Related links Group Type on page 594
Receive Space Disconnect
If enabled, the conversion resource can disconnect after receiving 1.6 seconds of space. Available only with an integrated type modem pool. Related links Group Type on page 594
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Modem Pool Group
Receiver Responds to Remote Loop
If enabled, the far-end modem can put conversion resource into loop back mode. Available only with an integrated type modem pool. Related links Group Type on page 594
Send Space Disconnect
If enabled, the conversion resource can send 4 seconds of space before disconnecting Available only with an integrated type modem pool. Related links Group Type on page 594
Speed
Valid Entry LOW (0 to 300 blind sampled) 300 1200 2400 4800 9600 19200
Usage
The communication speed in bits per second of the conversion resources in the group. One to three speeds are separated by slashes. For example, 300/1200/2400 indicates a maximum of three running speeds. Entry is required for a combined type modem pool.
Related links Group Type on page 594
Synchronization
The synchronization mode of the conversion resources in the group. Entry required for a combined type modem pool.
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Valid Entry sync async
Usage Synchronous Asynchronous
Related links Group Type on page 594
Time Delay
Time Delay
Available only with a combined type modem pool.
Valid Entry 0 to 255
Usage
The time delay in seconds to insert between sending the ringing to the modem and the off-hook alert to the data module.
Related links Group Type on page 594
CIRCUIT PACK ASSIGNMENTS
Circuit Pack Location
The port associated with the conversion resource on the integrated modem pool circuit pack. Available only with an integrated type modem pool.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
01 to 31 1 to 250 V1 to V9 01 to 31
Usage First and second characters are the cabinet number Third character is the carrier Fourth and fifth character are the slot number Six and seventh characters are the circuit number
Gateway Module Circuit
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Modem Pool Group
Example 01A0612 is in cabinet 01, carrier A, slot 06, and circuit number (port) 12. Related links Group Type on page 594
PORT PAIR ASSIGNMENTS
Analog Digital
The port numbers of the modem/TDM pair in a conversion resource. Two port entries are required. Available only with a combined type modem pool.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
01 to 31 1 to 250 V1 to V9 01 to 31
Usage First and second characters are the cabinet number Third character is the carrier Fourth and fifth character are the slot number Six and seventh characters are the circuit number
Gateway Module Circuit
Example 01A0612 is in cabinet 01, carrier A, slot 06, and circuit number (port) 12.
Related links Group Type on page 594
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Chapter 90: MOH Group
Use the MOH Group screen to define a collection of analog station and/or aux trunk port circuit pack ports that are connected to external audio sources for use with the Music on Hold feature. Example command: change moh-analog-group
MOH Source Location
The Music-on-hold analog or aux-trunk port location
Valid Entry 1 to 64 A to E 0 to 20
Usage First and second characters are the cabinet number Third character is the carrier Fourth and fifth character are the slot number
Table continues...
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MOH Group
Valid Entry 01 to 04 (Analog TIE trunks)
01 to 31 1 to 250 V1 to V9 01 to 31
Usage Six and seventh characters are the circuit number
Gateway Module Circuit
Group Name
The name that identifies the Music-on-hold (MOH) group.
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Chapter 91: Multifrequency-SignalingRelated Parameters
Sets the system or location parameters associated with multifrequency signaling.
Note: With the Multinational Locations feature enabled, you can assign MFC signal sets per trunk group, rather than system-wide. Example command: change multifrequency-signaling Related links Multinational Locations on page 838
Multifrequency-Signaling-Related Parameters: page 1
ANI Prefix
The prefix to apply to an extension when ANI is sent to the local telephone company central office. Accepts one to six digits.
Available only if the Outgoing Call Type is group-ii-mfc or mfe.
Related links Outgoing Call Type on page 605
Backward Cycle Timer (sec)
Available only if the Incoming Call Type is mfe.
Valid Entry 1 to 255
Usage
The number of seconds the system waits to send the check frequency after receiving an MFE signal.
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Multifrequency-Signaling-Related Parameters
Related links Incoming Call Type on page 603
Collect All Digits Before Seizure
Available only if the Outgoing Call Type is group-ii-mfc or mfe.
Valid Entry y
n
Usage
The system collects all the digits before seizing the trunk. Values administered for ANI Req for AAR and ARS Digit Conversion do not apply.
ANI collection is controlled by ARS administration.
Related links ANI Reqd Outgoing Call Type on page 605
Convert First Digit End-of-Dial To
The digit used when the incoming initial end-of-ani or end-of-dial MF signal is converted on a perswitch basis.
Available only if Private Group II Permissions and Public Interworking is enabled.
Related links Private Group II Permissions and Public Interworking on page 607
Default ANI
Available only if the Outgoing Call Type is group-ii-mfc or mfe.
Valid Entry 2 to 15 digits
blank
Usage
The switch or server identification number that is sent to the local telephone company central office (CO) when ANI is requested (by the CO) on a particular call but is unavailable, such as on tandem tie trunk calls.
Use for tandem switching. If blank, administer a value to send for ANI on outgoing calls when it is unavailable.
Related links ANI Not Available on page 612 Outgoing Call Type on page 605
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Multifrequency-Signaling-Related Parameters: page 1
Forward Cycle Timer (sec)
Available only if the Incoming Call Type is mfe.
Valid Entry 1 to 255
Usage
The number of seconds the system waits to receive the check frequency after sending an MFE signal. Communication Manager drops the call if the time runs out before it receives check frequency.
Related links Incoming Call Type on page 603
Group II Called Party Category
The type of Group II signals that should be used on the outgoing R2-MFC call. Available only if the Outgoing Call Type is group-ii-mfc and Use COR for All Group II Responses is disabled.
Valid Entry user-type
call-type
Usage
The type of telephone making the call determines the type of Group II signal that the switch or server sends (normal = ordinary telephone set, attendant = attendant console, data-call = data modules and similar data endpoints).
The dialed digits determine the type of Group II signal that the server sends.
Related links Outgoing Call Type on page 605 Use COR for All Group II Responses on page 608
Incoming Call Type
Valid Entry group-ii-mfc non-group-ii-mfc mfe
Usage The signal type that a local telephone company central office uses to place an incoming call to the server.
Multi-frequency Espanol used for Spain
Incomplete Dial Timer (sec)
Available only if the Incoming Call Type is mfe.
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Multifrequency-Signaling-Related Parameters
Valid Entry 45 to 255
Usage
The number of seconds Communication Manager waits from the start of a call until the end of the check frequency of the last signal. Communication Manager drops the call if the time runs out before it receives the check frequency.
Related links Incoming Call Type on page 603
Maintenance Call Type
Available only if the Incoming Call Type is group-ii-mfc or non-group-ii-mfc.
Valid Entry 1
2
none
Usage
The Belgium maintenance sequence is indicated when the local telephone company central office (CO) sends an MFC maintenance tone.
The Saudi Arabian sequence is indicated when the CO sends an MFC maintenance tone.
Not administered.
Related links Incoming Call Type on page 603
Maximum Resend Requests
The threshold number of resend type MFC signals the server running Communication Manager accepts during an outgoing call.
Valid Entry 1 to 99 1 blank
Usage The number of resend requests. The call is dropped if one resend signal is received. An unlimited number of resend requests is allowed.
MF Signaling Intercept Treatment - Incoming
Valid Entry y
n
Usage
Sends the group B signal for the intercept to the local telephone company central office and play intercept tone on the trunk.
Uses normal DID/TIE/ISDN intercept treatment.
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Multifrequency-Signaling-Related Parameters: page 1
MF Signaling Intercept Treatment - Outgoing
Defines the treatment for outgoing calls that cannot be completed as dialed. Available only if Outgoing Call Type is group-ii-mfc.
Valid Entry announcement
tone
Usage
Plays a recorded announcement for outgoing calls that cannot be completed as dialed. You select and record the message. Requires entry of the extension number for the announcement.
Plays intercept tone for outgoing calls that cannot be completed as dialed.
Related links Outgoing Call Type on page 605
MFE Type
Available only if the Incoming Call Type is mfe.
Valid Entry 2/5 2/6
Usage Determines which public signaling Communication Manager uses.
Related links Incoming Call Type on page 603
Outgoing Call Type
Valid Entry group-ii-mfc
mfe none
Usage
The signal type that the switch or server uses to place an outgoing call to a local telephone company central office.
Multi-frequency Espanol used in Spain.
Not administered.
Outgoing Forward Signal Absent Timer (sec)
Available only if the Outgoing Call Type is group-ii-mfc.
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Multifrequency-Signaling-Related Parameters
Valid Entry 11 to 255
Usage
The maximum time to elapse between forward signals on outgoing calls. The timer starts and restarts when a forward tone is taken off the link, and it stops when the next forward tone is applied to the link.
Related links Outgoing Call Type on page 605
Outgoing Forward Signal Present Timer (sec)
Available only if the Outgoing Call Type is group-ii-mfc.
Valid Entry 1 to 255
Usage
The maximum time to elapse between signals on a call. This timer runs when MFC tones are being sent or received on an outgoing call. The timer starts (and restarts) when Communication Manager begins sending a forward signal and stops when Communication Manager receives the backward signal.
Related links Outgoing Call Type on page 605
Outgoing Start Timer (sec)
Available only if the Incoming Call Type is mfe.
Valid Entry 1 to 255
Usage
The number of seconds from seizure until the beginning of the first Group A signal from the receiving end, and from the end of the check frequency until the beginning receipt of the first digit following the Group II signal.
Related links Incoming Call Type on page 603
Overlap Sending on Link-to-Link Tandem Calls
If enabled, and calls are tandemed between servers, then ANI for the switch or server is sent to the terminating switch if that switch requests ANI before Communication Manager receives it from the originating server or switch. The terminating server or switch can request ANI before the receipt of the last address digit if it is not running Communication Manager, or if Communication Manager is administered to request the call category at the start of the call.
If enabled, Communication Manager sends and receives digits one digit at a time instead of enbloc. With enbloc, digits are not sent until the entire group of digits is received.
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Multifrequency-Signaling-Related Parameters: page 1
Available only if Collect All Digits Before Seizure is disabled. Related links
Collect All Digits Before Seizure on page 602 Request Call Category at Start of Call on page 611
Private Group II Permissions and Public Interworking
If enabled, then Communication Manager:
· Sends the category for MFC ANI for the COR of the originating party for non-private-MFCtrunk to MFC-private-trunk calls.
· Sends the Group II category received over the incoming private trunk as the outgoing Group II category on tandem private MFC calls.
· Applies MFC group II-CPC termination restrictions on incoming MFC private trunk calls.
· Checks station permissions for call forward off-net calls.
Available only if the Incoming Call Type is group-ii-mfc or non-group-ii-mfc and the Outgoing Call Type is group-ii-mfc or none.
Related links Incoming Call Type on page 603 Outgoing Call Type on page 605
Received Signal Gain (dB)
Valid Entry -15 to 3
Usage
The number for the loss or gain when the MFC port listens to the trunk port. Communication Manager listens with a range of -5 to -35. This value moves the range (for example, a value of -5 provides a range of -10 to -40). Also applies to Russian MF Shuttle trunks.
Request Incoming ANI (non-AAR/ARS)
If enabled, ANI is requested on incoming R2-MFC calls.
Available only if Incoming Call Type is group-ii-mfc or mfe and the Outgoing Call Type is groupii-mfc or mfe. Applies only if the incoming call via the R2-MFC trunk is terminating to a local station on this PBX.
Related links Incoming Call Type on page 603 Outgoing Call Type on page 605
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Multifrequency-Signaling-Related Parameters
Transmitted Signal Gain (dB)
Valid Entry -15 to 3
Usage
The number for the loss or gain when the trunk port listens to the MFC port. The MFC port generates at -5 for MFC and -8 for MFE. This field adds gain or loss to the starting value of -5. Also applies to Russian Shuttle trunks and Russian multifrequency ANI.
Use COR for All Group II Responses
If enabled, the COR administered category is used for both the calling party and called party categories.
Available only if the Outgoing Call Type is group-ii-mfc.
Related links Outgoing Call Type on page 605
Use COR for Calling Party Category
Indicates the category to send with ANI if requested on an outgoing R2-MFC call. Available only if the Outgoing Call Type is group-ii-mfc and Use COR for All Group II Responses is disabled.
Valid Entry y n
Usage The calling facility's COR is used to determine category. The calling party's user-type COR is used to determine category.
Related links Outgoing Call Type on page 605 Use COR for All Group II Responses on page 608
NEXT ANI DIGIT
Related links Incoming Call Type on page 603 Outgoing Call Type on page 605
Incoming
Available only if the Incoming Call Type is group-ii-mfc and the Outgoing Call Type is group-iimfc or mfe.
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Multifrequency-Signaling-Related Parameters: page 2
Valid Entry next-digit next_ani_digit send-ani
Usage
Determines whether the Next ANI Digit signal is the same as the send-ani signal, the next-digit signal, or another signal defined as next_ani_digit.
Outgoing
Available only if the Outgoing Call Type is group-ii-mfc.
Valid Entry next-digit next_ani_digit send-ani
Usage
Determines whether the Next ANI Digit signal is the same as the send-ani signal or the next-digit signal or another signal defined as next_ani_digit.
Related links Outgoing Call Type on page 605
Multifrequency-Signaling-Related Parameters: page 2
These fields define call category and ANI information. For India, the ANI can be requested without the call category information.
Address Digits Include End-of-Digits Signal
If enabled, indicates that an outgoing forward Group I end-of-digit signal is always sent after completion of address digits upon request from the local telephone company central office for outgoing calls.
ANI Source for Forwarded & Covered Calls
Valid Entry caller forwarder
Usage Send the calling party's ANI when calls are redirected. Send the forwarding party's ANI when calls are redirected.
Call Category for Vector ii-digits
If enabled, allows the use of the call category digit, which is part of ANI, as the ii-digits on call vector steps.
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Multifrequency-Signaling-Related Parameters
Do Not Send Group B Signals to CO
Available only if the Incoming Call Type is group-ii-mfc.
Valid Entry y n
Usage Does not send Group-B signals to complete an incoming call. Sends Group-B signals to complete an incoming call.
Related links Incoming Call Type on page 603
Number of Incoming ANI Digits
Valid Entry 0 to 15
Usage The number of ANI digits for incoming MFC calls.
Number of Outgoing ANI Digits
Valid Entry 0 to 15
Usage
The number of ANI digits for outgoing MFC calls.
In India or any country where end-of-ani and end-of-digits are not defined for Tones to CO on Outgoing Forward Calls - Group I, Avaya Communication Manager appends ANI-Not-Available digits to ANI digits if the actual ANI length is less than the number entered in this field. If end-of-ani or end-of-digits are defined, this field is used in conjunction with Truncate Station Number in ANI as a maximum ANI length. For India, even if the length of ANI is defined, if the timeout occurs during the ANI collection, the call is routed with the ANI digits already collected.
Applies to Russian shuttle trunks, and MFC and MFE trunks.
Outgoing II by COR
Available only if either Use COR for Calling Party Category or Use COR for All Group II Responses are enabled.
Valid Entry 1 to 10
Usage
The Group II signal sent to the local telephone company central office on outgoing calls can be administered per COR (Class of Restriction) and per trunk group. The Group II signal is administered per COR. That per-COR value in turn can be mapped into a different outgoing signaling parameter set. The values for this field administer that outgoing mapping.
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Multifrequency-Signaling-Related Parameters: page 2
Related links Use COR for All Group II Responses on page 608 Use COR for Calling Party Category on page 608
Request Call Category at Start of Call
Indicates that the Send-ANI backward signal requesting for the caller-category information is sequenced differently in the MFC signaling flow. The Caller-category Request backward signal is disjointed from the ANI request. If enabled, the Send-ANI backward signal corresponds exclusively to the caller-category request. In response to this signal, Communication Manager sends a forward signal containing the callercategory information on outgoing calls. On incoming calls, Communication Manager sends the Send-ANI backward signal upon receipt of the first address signal.
Request CPN at Start of Call
With Request CPN at Start of Call, Communication Manager can collect ANI and call category immediately after receipt of the first address digit. If enabled, provides ANI Calling Party Number (CPN) and call category immediately after receiving the first address digit. Available only if the Incoming Call Type is group-ii-mfc. Related links Incoming Call Type on page 603
Restart ANI from Caller Category
If enabled, Avaya Communication Manager sends the caller-category signal later again when the signals for Caller-Category and ANI requests are the same and this signal is received after the Next-Digit forward signals have been received.
Truncate Station Number in ANI
Applies to Russian shuttle trunks, and MFC and MFE trunks.
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Multifrequency-Signaling-Related Parameters
Valid Entry beginning ending no
Usage
Defines the side of the extension number from which to truncate when station ANI is sent to the local telephone company central office and the combined length of the ANI prefix and extension number is greater than the administered Number of Outgoing ANI Digits. The ANI prefix (either MFC or COR) is not truncated. There is no effect if ANI for switch or server is sent.
Related links Number of Outgoing ANI Digits on page 610
INCOMING / OUTGOING
ANI Available
Valid Entry 1 to 15 blank
Usage The signal number used for incoming ANI-Available.
ANI Not Available
Required if the Default ANI is not administered.
Valid Entry 1 to 15 blank
Usage
The signal number used for outgoing ANI-Available. Communication Manager outpulses the End-of-Dial backward signal when the ANI-Not-Available forward signal is received on incoming calls. Communication Manager outpulses the ANINot-Available forward signal to the local telephone company central office on outgoing calls where ANI is not possible.
Multifrequency-Signaling-Related Parameters: page 3
These fields define the meaning of MFC tones for calls originated at the local telephone company central office. When the system initially displays the screen, either of two sets of default values is possible. One set is for the Group II call type; the other set is for non-Group II call type. In each set, the default value for each field is set to the most common value. Available only if the Incoming Call Type is group-ii-mfc or non-group-ii-mfc. Related links Incoming Call Type on page 603
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Multifrequency-Signaling-Related Parameters: page 3
INCOMING FORWARD SIGNAL TYPES (Tones from CO)
Group I
Displays message codes 11 to 15. (Numbers 1 through 10 are assigned to the digits of the destination telephone number.) A Group I signal type can be administered for each code.
Group I signals are a set of forward signals generated by the originating server.
Valid Entry ani-avail
ani-not-avail
drop end-of-ani
end-of-dial ignored
maint-call
Usage
Used in Hungary. If this signal is defined and Automatic Number Identification (ANI) is requested on outgoing R2-MFC calls, ANI is sent to the local telephone central office before ANI caller digits are sent. This signal is sent after the ANI caller category signal.
Available if Incoming Call Type is group-ii-mfc.
Used on Direct Outward Dialing calls in Brazil and Columbia. Communication Manager sends this signal to the local telephone company central office when it receives an Automatic Number Identification (ANI) request and the caller's number is unavailable.
Available if Incoming Call Type is group-ii-mfc.
When this signal is received from the local telephone company central office, Avaya Communication Manager starts the disconnect sequence and drops the call.
Available if Incoming Call Type is group-ii-mfc or non-group-ii-mfc.
This signal is used on Direct Outward Dialing and Direct Inward Dialing calls. Communication Manager sends this signal to indicate the end-of-ANI digits when Automatic Number Identification (ANI) digits are sent to the local telephone company central office.
Available if Incoming Call Type is group-ii-mfc.
This signal is used when open numbering is used on Direct Inward Dialing calls. The local telephone company central office sends this signal to indicate the end-of-dial digits and Communication Manager responds with a request for a Group II signal.
Available if Incoming Call Type is group-ii-mfc.
If this signal is received from the local telephone company central office, Communication Manager sends a corresponding signal (A.1, and so on) but no action is taken in the response and it is not counted as a digit. In Belgium, this signal is not acknowledged.
Available if Incoming Call Type is group-ii-mfc or non-group-ii-mfc.
The local telephone company central office (CO) sends a signal to indicate that a call is a maintenance call and Communication Manager prepares the special maintenance call sequences for the CO. This signal can be used on Direct Inward Dialing calls in Saudi Arabia.
Available if Incoming Call Type is group-ii-mfc.
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Multifrequency-Signaling-Related Parameters
Valid Entry send-congest
Usage
When Communication Manager receives this signal from the local telephone company central office (CO) on a Direct Inward Dialing call, it returns a congestion signal (Group A), in compel (not pulse) mode, to the CO.
Available if Incoming Call Type is group-ii-mfc.
Group II
Displays message codes 1 to 15. A Group II signal type can be administered for each code. Group II signals are a more elaborate set of forward signals generated by the originating server.
Valid Entry attendant
busy-rt-attd data-call
data-verify
drop maint-call normal
send-intercept
Usage
If Communication Manager receives this signal on Direct Inward Dialing (DID) calls, the call terminates at an attendant regardless of the extension dialed. On Direct Outward Dialing (DOD) calls, this signal is sent to the local telephone company central office (CO) if the CO requests calling-category information and the originating extension is an attendant. This signal is used on both DID and DOD calls.
If Communication Manager receives this signal on Direct Inward Dialing (DID) calls, the call terminates at an attendant if the called extension is busy or at the called extension if it is not busy. This signal is used on DID calls.
If Communication Manager receives this signal on Direct Inward Dialing (DID) calls, it sends intercept treatment. On Direct Outward Dialing (DOD) calls, this signal is sent to the local telephone company central office (CO) if the CO requests callingcategory information and the originating extension is a data extension. This signal is used on both DID and DOD calls.
If Communication Manager receives this signal on Direct Inward Dialing (DID) calls and the terminating extension is not a data extension, it sends intercept treatment. On Direct Outward Dialing (DOD) calls, this signal is sent to the local telephone company central office (CO) if the CO requests calling-category information and the originating extension is a data extension. This signal is used on both DID and DOD calls.
When this signal is received from the local telephone company central office (CO), Communication Manager starts the disconnect sequence and drops the call.
The local telephone company central office (CO) sends a signal to indicate that a call is a maintenance call and Avaya Communication Manager prepares the special maintenance call sequences for the CO.
This signal indicates that the caller is a normal subscriber. If it is received on a Direct Inward Dialing (DID) call, the call is terminated at the called extension. For an outgoing MF signaling call that uses Group II signaling, this signal is sent to the local telephone company central office (CO) when the CO requests calling-category information and the originating extension is a station. This signal is used in both DID and Direct Outward Dialing calls.
If Communication Manager receives this signal from the local telephone company central office (CO) on a Direct Inward Dialing call, it returns Group B intercept signal to the CO.
Table continues...
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Valid Entry toll-auto
toll-operator
Multifrequency-Signaling-Related Parameters: page 3
Usage This signal is used in China. This signal indicates that a call is an automatic toll call. When the call terminates at a busy station and a special busy signal is defined, the busy signal is sent to the local telephone company central office. A special busy signal can be defined by choosing the option toll-busy on the incoming Group B signals. This signal, used in China, is treated as a normal subscriber signal. See the normal definition.
INCOMING BACKWARD SIGNAL TYPES (Tones to CO)
Group A
Displays message codes 11 to 15. (Numbers 1 through 10 are assigned to the digits of the destination telephone number.) A Group A signal type can be administered for each code.
Group A signals are backward signals generated by the destination server or switch.
Valid Entry congestion
end-of-dial
intercept
next-ani-digit next-digit send-ani
setup-sppath
Usage
The local telephone company central office sends this signal to indicate that it is experiencing network congestion. When Communication Manager receives this signal on Direct Outward Dialing (DOD) calls, it drops the trunk and plays reorder tone to the calling party. This signal is used on DOD calls.
This signal is sent to indicate the end of the address digit string. For MF Group II calls, this signal requests a Group II signal and switches the sender over to the Group B signaling mode. This signal is used on both Direct Inward Dialing and Direct Outward Dialing calls.
The local telephone company central office sends this signal to indicate the call has been terminated to an invalid destination. When Communication Manager receives this signal on Direct Outward Dialing (DOD) calls, it drops the trunk and plays intercept tone to the calling party. This signal is used on DOD calls.
Communication Manager sends this signal to request the next ANI digit. This signal is used on Direct Inward Dialing and Direct Outward Dialing calls.
Communication Manager sends this signal to request the next digit. This signal is used on both Direct Inward Dialing and Direct Outward Dialing calls.
The local telephone company central office (CO) sends this signal to request callingparty category and sends additional signals to request ANI digits. This signal is sent to the CO when Avaya Communication Manager requests ANI digits on Direct Inward Dialing (DID) calls. This signal is used on both Direct Outward Dialing and DID calls.
The local telephone company central office sends this signal to Communication Manager to set up a speech path. This signal is used on Direct Outward Dialing calls and on Direct Inward Dialing calls in Belgium.
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Multifrequency-Signaling-Related Parameters
Group B
Displays message codes between 1 and 15. A Group B signal type can be administered for each code
Group B signals enhance Group A signals for backward signaling from the destination end by providing the status of the called party. In addition, if the originating server uses Group II signals, the destination end answers with Group B signals.
Not available if Do Not Send Group B Signals to CO is enabled.
Valid Entry busy congestion free intercept mct
tariff-free tie-free toll-busy
Usage
This signal is sent to indicate that the called party is busy. On Direct Inward Dialing calls, the signal is sent to the local telephone company central office if there is no coverage point to terminate the call. If Communication Manager receives this signal on Direct Outward Dialing calls, it plays busy tone to the calling party and drops the trunk.
This signal is sent to indicate that the system is congested and the call cannot be terminated successfully. On Direct Inward Dialing calls, the signal is sent to the local telephone company central office to indicate that a resource is not available. On Direct Outward Dialing calls, if Communication Manager receives this signal, reorder tone is played to the calling party and the trunk is dropped.
This signal indicates that the called party is idle. On Direct Inward Dialing calls, the signal is sent to the local telephone company central office to indicate that the called party is idle and the call is terminated successfully. If Communication Manager receives this signal on Direct Outward Dialing calls, it connects the trunk to the calling party.
This signal indicates that the called party number is not in service or is incorrect. On Direct Inward Dialing calls, if intercept treatment is set to provide a tone, tone is sent to the local telephone company central office to indicate that the called number is invalid. If Communication Manager receives the signal on Direct Outward Dialing calls, it plays intercept tone to the calling party and drops the trunk.
This signal identifies the call as one that needs to be traced by the local telephone company central office (CO). Avaya Communication Manager then generates an MFC Call Trace Backward Signal administered for multifrequency signaling during call setup instead of the "free" signal. If the terminating station's Class of Restriction (COR) is enabled, the CO collects trace information before releasing the calling party. If the station's COR has MF Incoming Call Trace enabled and the "mct" signal is not defined, then the "free" signal is sent.
This signal is sent when the trunk group provides an 800 service. Avaya Communication Manager generates an MFC tariff-free backward signal during call setup instead of the "free" signal, facilitating local telephone company central office billing. If the trunk is administered as a tariff-free trunk and the "tariff-free" signal is not defined, then the "free" signal is sent.
This signal is used only when an incoming call is received and defined and the incoming facility is a tie trunk. Otherwise, the free signal is used.
This signal, used in China, is sent to indicate that the called party is busy if the call is an automatic toll call.
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Multifrequency - Signaling- Related Parameters: page 4
Related links Do Not Send Group B Signals to CO on page 610
Multifrequency - Signaling- Related Parameters: page 4
The fields shown on this page define the meaning of MFC tones for calls originated at the switch or server. The system displays this screen only if Outgoing Call Type is group-ii-mfc or mfe. Related links Outgoing Call Type on page 605
OUTGOING FORWARD SIGNAL TYPES (Tones to CO)
Group I
Displays message codes 11 to 15. (Numbers 1 through 10 are assigned to the digits of the destination telephone number.) A Group I signal type can be administered for each code.
Group I signals are a set of forward signals generated by the originating server.
Valid Entry ani-avail
ani-not-avail
end-of-ani
end-of-digits
Usage
Used in Hungary. If this signal is defined and Automatic Number Identification (ANI) is requested on outgoing R2-MFC calls, ANI is sent to the local telephone central office before ANI caller digits are sent. This signal is sent after the ANI caller category signal.
Used on Direct Outward Dialing calls in Brazil and Columbia. Communication Manager sends this signal to the local telephone company central office when it receives an Automatic Number Identification (ANI) request and the caller's number is unavailable.
This signal is used on Direct Outward Dialing and Direct Inward Dialing calls. Communication Manager sends this signal to indicate the end-of-ANI digits when Automatic Number Identification (ANI) digits are sent to the local telephone company central office.
This signal is sent by the originating server that makes outgoing calls, sends digits, and receives a next-digit Group A signal from the destination server or switch when there are no more digits to be sent. This signal is also sent when Communication Manager does not have end-of-ani assigned, makes an outgoing call, sends ANI, and receives a call-info-ani Group A signal from the destination end when there are no more Automatic Number Identification (ANI) digits to be sent. If both end-of-digits and end-of-ani are assigned, Communication Manager uses end-of-ani after it sends the last ANI digit and end-of-digits after sending the last called-number digit.
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Multifrequency-Signaling-Related Parameters
Group II
Displays message codes between 1 and 15. A Group II signal type can be administered for each code. Each entry can only appear once in the Group II column.
Group II signals are a more elaborate set of forward signals generated by the originating server.
Valid Entry attendant data-call normal
toll-auto
Usage
If Communication Manager receives this signal on Direct Inward Dialing (DID) calls, the call terminates at an attendant regardless of the extension dialed. On Direct Outward Dialing (DOD) calls, this signal is sent to the local telephone company central office (CO) if the CO requests calling-category information and the originating extension is an attendant. This signal is used on both DID and DOD calls.
If Communication Manager receives this signal on Direct Inward Dialing (DID) calls, it sends intercept treatment. On Direct Outward Dialing (DOD) calls, this signal is sent to the local telephone company central office (CO) if the CO requests callingcategory information and the originating extension is a data extension. This signal is used on both DID and DOD calls.
This signal indicates that the caller is a normal subscriber. If it is received on a Direct Inward Dialing (DID) call, the call is terminated at the called extension. For an outgoing MF signaling call that uses Group II signaling, this signal is sent to the local telephone company central office (CO) when the CO requests calling-category information and the originating extension is a station. This signal is used in both DID and Direct Outward Dialing calls.
This signal is used in China. This signal indicates that a call is an automatic toll call. When the call terminates at a busy station and a special busy signal is defined, the busy signal is sent to the local telephone company central office. A special busy signal can be defined by choosing the option toll-busy on the incoming Group B signals.
OUTGOING BACKWARD SIGNAL TYPES (Tones from CO)
Group A
Displays message codes between 1 and 15. A Group A signal type can be administered for each code.
Group A signals are backward signals generated by the destination server or switch.
Valid Entry congestion
drop
Usage
The local telephone company central office sends this signal to indicate that it is experiencing network congestion. When Communication Manager receives this signal on Direct Outward Dialing (DOD) calls, it drops the trunk and plays reorder tone to the calling party. This signal is used on DOD calls.
When this signal is sent, the receiving end starts the disconnect sequence.
Table continues...
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Multifrequency - Signaling- Related Parameters: page 4
Valid Entry end-of-dial
intercept
last-2-digits last-3-digits last-digit next-ani-digit next-digit restart resend-digit send-ani
setup-sppath
Usage
This signal is sent to indicate the end of the address digit string. For MF Group II calls, this signal requests a Group II signal and switches the sender over to the Group B signaling mode. This signal is used on both Direct Inward Dialing and Direct Outward Dialing calls.
The local telephone company central office sends this signal to indicate the call has been terminated to an invalid destination. When Communication Manager receives this signal on Direct Outward Dialing (DOD) calls, it drops the trunk and plays intercept tone to the calling party. This signal is used on DOD calls.
Communication Manager sends this signal to adjust the outpulsing pointer so that the last three digits can be resent. This signal is used on Direct Outward Dialing calls.
Communication Manager sends this signal to adjust the outpulsing pointer so that the last four digits can be resent. This signal is used on Direct Outward Dialing calls.
Communication Manager sends this signal to adjust the outpulsing pointer so that the last two digits can be resent. This signal is used on Direct Outward Dialing calls.
Communication Manager sends this signal to request the next ANI digit. This signal is used on Direct Inward Dialing and Direct Outward Dialing calls.
Communication Manager sends this signal to request the next digit. This signal is used on both Direct Inward Dialing and Direct Outward Dialing calls.
Communication Manager sends this signal to request the whole digit string again. This signal is used on Direct Outward Dialing calls.
Communication Manager sends this signal to adjust the outpulsing pointer so that the last digit can be resent again. This signal is used on Direct Outward Dialing calls.
The local telephone company central office (CO) sends this signal to request callingparty category and sends additional signals to request ANI digits. This signal is sent to the CO when Avaya Communication Manager requests ANI digits on Direct Inward Dialing (DID) calls. This signal is used on both Direct Outward Dialing and DID calls.
The local telephone company central office sends this signal to Communication Manager to set up a speech path. This signal is used on Direct Outward Dialing calls and on Direct Inward Dialing calls in Belgium.
Group B
Displays message codes between 1 and 15. A Group B signal type can be administered for each code.
Group B signals enhance Group A signals for backward signaling from the destination end by providing the status of the called party. In addition, if the originating server uses Group II signals, the destination end answers with Group B signals.
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Multifrequency-Signaling-Related Parameters
Valid Entry busy congestion free intercept mct
tariff-free toll-busy
Usage
This signal is sent to indicate that the called party is busy. On Direct Inward Dialing calls, the signal is sent to the local telephone company central office if there is no coverage point to terminate the call. If Communication Manager receives this signal on Direct Outward Dialing calls, it plays busy tone to the calling party and drops the trunk.
This signal is sent to indicate that the system is congested and the call cannot be terminated successfully. On Direct Inward Dialing calls, the signal is sent to the local telephone company central office to indicate that a resource is not available. On Direct Outward Dialing calls, if Communication Manager receives this signal, reorder tone is played to the calling party and the trunk is dropped.
This signal indicates that the called party is idle. On Direct Inward Dialing calls, the signal is sent to the local telephone company central office to indicate that the called party is idle and the call is terminated successfully. If Communication Manager receives this signal on Direct Outward Dialing calls, it connects the trunk to the calling party.
This signal indicates that the called party number is not in service or is incorrect. On Direct Inward Dialing calls, if intercept treatment is set to provide a tone, tone is sent to the local telephone company central office to indicate that the called number is invalid. If Communication Manager receives the signal on Direct Outward Dialing calls, it plays intercept tone to the calling party and drops the trunk.
This signal identifies the call as one that needs to be traced by the local telephone company central office (CO). Avaya Communication Manager then generates an MFC Call Trace Backward Signal administered for multifrequency signaling during call setup instead of the "free" signal. If the terminating station's Class of Restriction (COR) is enabled, the CO collects trace information before releasing the calling party. If the station's COR has MF Incoming Call Trace enabled and the "mct" signal is not defined, then the "free" signal is sent.
This signal is sent when the trunk group provides an 800 service. Avaya Communication Manager generates an MFC tariff-free backward signal during call setup instead of the "free" signal, facilitating local telephone company central office billing. If the trunk is administered as a tariff-free trunk and the "tariff-free" signal is not defined, then the "free" signal is sent.
This signal, used in China, is sent to indicate that the called party is busy if the call is an automatic toll call.
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Chapter 92: Multiple Level Precedence & Preemption (MLPP) Parameters
Use this screen to set up system parameters for the Multiple Level Precedence & Preemption feature. Example command: change system-parameters mlpp
ANNOUNCEMENTS
ASAC Enabled
Note:
The ASAC Enabled field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
ASAC is an extension to the existing Multiple Level Precedence and Preemption (MLPP) calling function which provides directionalization (inbound and outbound awareness) attribute to a calling budget. Administrators can restrict the available call capacity in the inbound and outbound direction with ASAC.
This field is available only when you set the Multiple Level Precedence and Preemption field on the System Parameters Customer Options screen to y.
Valid entry y
n
Usage
Makes all ASAC-related fields available on the SIP Trunk Group screen. ASAC also allows Destination Code Control (DCC) functionality.
Makes ASAC-related fields unavailable on the SIP Trunk Group screen. Therefore, the ASAC functionality is unavailable for calls. The default value is n.
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Multiple Level Precedence & Preemption (MLPP) Parameters
Attendant Queue Announcement
Note: This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers. The extension of the attendant queue announcement for incoming precedence calls that are above the Routine precedence level and are placed in the attendant waiting queue. To enable Attendant Queue Announcement, you must set the Annc Type field on the Announcements/Audio Sources screen to Integrated.
Blocked Precedence Level
The extension of the Blocked Precedence Level announcement.
Busy, Not Equipped
The extension of the Busy, Not Equipped for Preemption announcement.
Service Interruption
The extension of the Service Interruption announcement.
Unauthorized Precedence Level
The extension of the Unauthorized Precedence Level announcement.
Vacant Code
The extension of the Vacant Code announcement.
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PRECEDENCE CALLING-DIALED DIGIT ASSIGNMENT
PRECEDENCE CALLING-DIALED DIGIT ASSIGNMENT
Caution: Do not change the default Precedence Calling dialed digits unless you are coordinating this change with other companion networks in your system. If the Precedence Calling digits are different across networks, the system does not properly process the calls. Each of the Precedence Calling digits must be different. You cannot use the same digit for two different precedence levels.
Attendant Diversion Timing (sec)
Valid Entry 10 to 99 blank
Usage
Controls how many seconds this type of call rings before the call is routed to the Remote Attendant Route String that consists of any valid telephone number on the network. Attendant Diversion Timing is usually a backup answering position for the remote attendant console. The Remote Attendant Route String does not raise the precedence level of the call.
Default Route Digit
Available only if Worldwide Numbering Dial Plan is enabled. A valid digit is required in this field.
Valid Entry 0 1 2 3 4 5 6 7 8 9
Usage Voice call (the default value) Circuit switched data call Satellite avoidance call (reserved) (reserved) Hotline voice grade call Hotline data grade call (reserved) (reserved) (reserved)
Related links Worldwide Numbering Dial Plan Active on page 626
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Multiple Level Precedence & Preemption (MLPP) Parameters
Default Service Domain
Valid Entry 0 to 16777215
Usage
The system service domain number. This number must be unique within a switching network. The system uses the system service domain to determine eligibility for precedence calling when interswitch precedence calls over non-ISDN trunks occur.
Flash
The digit assignment for Flash precedence level calls. The default is 1.
Flash Override
The digit assignment for Flash Override precedence level calls. The default is 0.
Immediate
The digit assignment for Immediate precedence level calls. The default is 2.
ISDN Precedence Call Timeout (sec)
Valid Entry 4 to 30
Usage
The timeout seconds used instead of the Precedence Call Timeout when the call is from a Multiple Level Precedence and Preemption (MLPP) ISDN-PRI trunk. Default is 30 seconds.
Line Load Control Restriction Level
Determines what stations, based on their COR, are restricted from originating calls.
Valid Entry 0 2 3 4
Usage Feature not active (no restrictions). This is the default. Restrict stations with a COR assigned to LLC levels 2, 3, and 4. Restrict stations with a COR assigned to LLC levels 3 and 4. Restrict stations with a COR assigned to LLC level 4.
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PRECEDENCE CALLING-DIALED DIGIT ASSIGNMENT
Precedence Call Timeout (sec)
Valid Entry 4 to 30
Usage
The number of seconds a precedence call remains in call waiting status before it is diverted. A busy user receives a precedence call waiting tone only if the incoming call cannot be connected and cannot preempt the user. The called party hears the tone every 10 seconds until answered or the administered time-out occurs. If ignored, the caller is diverted to an attendant or a call-forwarded station. The default is 30 seconds.
Preempt Emergency Call
If enabled, allows preemption of an Emergency 911 call made from a preemptable station by a higher precedence call.
Note:
Emergency call preemption occurs only in case of trunk and endpoint preemption. In case of network preemption the emergency calls are not preempted, irrespective of the settings for Preempt Emergency Call? flag.
Priority
The digit assignment for Priority precedence level calls. The default is 3.
Remote Attendant Route String
A user-defined telephone to which a precedence call can be routed when no console or night telephone is administered. Accepts from 1 to 24 digits.
Routine
The digit assignment for Routine precedence level calls. The default is 4.
WNDP Emergency 911 Route String
This route string is outpulsed when a user dials either 911 and waits for the interdigit timeout, or dials 911 followed by #. This dialing option only works when the WNDP Flash FAC is 91. Use a trunk access code (TAC), the AAR or the ARS access code, a WNDP access code, or an
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Multiple Level Precedence & Preemption (MLPP) Parameters
extension. For a WNDP access code, use the access code for the lowest precedence calling level in the system. Accepts from 1 to 24 digits.
Note: An Emergency/911 call is a call that routes using the ARS table with the call type defined as either "alrt" or "emer".
Worldwide Numbering Dial Plan Active
Enables or disables the Worldwide Numbering Dial Plan. Disabled by default.
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Chapter 93: Music Sources
Use this screen to define music sources for Tenant Partitions. Each music source defined on this screen can be used by one or more Tenant Partitions. However, a partition can have only one music source.
Note: If you use equipment that rebroadcasts music or other copyrighted materials, you might be required to obtain a copyright license from, or pay fees to, a third party. You can purchase a Magic Hold system, which does not require such a license, from Avaya Inc. or Avaya Business Partners. Available only if Tenant Partitioning is enabled for the system. Example command: change music-sources
Related links Tenant Partitioning on page 840
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Music Sources
Description
A description of the administered music source. Accepts up to 20 alphanumeric characters. Available only if music or tone is administered for treatment for hold. Related links Type (Column) on page 628
Source
Available only if music is administered for treatment for hold.
Valid Entry ext group port
Usage Audio source extension for a single or group audio source. A Music-on-Hold analog group number. An analog or auxiliary trunk source location.
Related links Type (Column) on page 628
Source No
The number assigned to this music source.
Type (Column)
Valid Entry music tone none
Usage The type of treatment to be provided by the music source. Only one music source can use the tone value.
Type (field)
Available only if music is administered for treatment for hold.
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Type (field)
Valid Entry ext group port
Usage The source for the music on hold. The source can be an announcement extension, an audio group, or a port on a VAL board.
Note:
A source identifier (extension number, audio group number, or port number) must be administered with the source type.
Related links Type (Column) on page 628
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Chapter 94: Network Facilities
Used to administer new network-provided service or feature names and corresponding ISDN PRI (network specific facilities information element) encodings, for call-by-call trunk groups. Values for pre-defined facilities are displayed at the top of the screen and are display-only. User-defined facilities and services can be entered in the fields below. When Usage Allocation Enhancements is enabled for the system, the system displays the Network Facilities screen. Additional user-defined entries can be administered on this screen. Example command: change isdn network-facilities
Facility Coding
The ISDN-specified value for this service or feature.
Facility Type
The facility type. For types 2 and 3, Usage Allocation Enhancements must be enabled for the system.
Valid Entry 0 - feature 1 - service 2 - incoming 3 - outgoing
Usage Predefined features. Predefined services. An incoming-type user-defined entry. An outgoing-type user-defined entry.
Related links Usage Allocation Enhancements on page 841
Name
The name for the feature or service.
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Chapter 95: Network Region Group
A network region group is a collection of network regions that share one network connection to the core and have infinite bandwidth between them. A network region group is used for bandwidth management. To administer the fields on the Network Region Group screen, use the following command: change network-region-group n, where n is the network region group number.
Name
Use this field to assign a unique name to the network region group. You can enter up to 20 alphanumeric characters. The name must not start or end with a blank space, but can contain internal blank spaces. The name is case-sensitive and must not contain the following characters: <>^%$@#*!'?. This name must exactly match a location name administered through System Manager on the Elements > Routing > Locations page. Do not set this field unless the BW Management Option field is set to shared-SM.
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Chapter 96: Node Number Routing
Specifies the routing pattern associated with each node in a public or private network. Node Number Routing is a required capability for Extension Number Portability (ENP) and is associated with the Uniform Dial Plan (UDP). Example command: change node-routing n, where n is the node number.
Node Number
The node number.
Partitioned Group Number
The partitioned group number associated with the node numbers being administered.
Route Pat
Valid Entry
Usage
1 to 2000 or blank The routing pattern associated with the corresponding node number.
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Chapter 97: Numbering-Private Format
Use this screen to specify the digits to be inserted in the calling number Information Element (IE), the connected number IE, and the QSIG party number for extensions in the Private Numbering Plan. Communication Manager can generate private network numbers up to 15 digits long. Communication Manager can receive and display private network numbers up to 21 digits long. Use the Numbering-Private Format screen to:
· Administer the extensions of all calls traversing the SIP trunks. · Configure the private numbering table. · Populate the calls to extensions terminating on a SIP-integrated Modular Messaging. To change the value of the fields on the Numbering-Private Format screen, use the following command: change private-numbering n, where n is the length of the extension. The command change private-numbering 0 starts at the first entry.
Ext Code
In the case of a four-digit Ext Len, an Ext Code of 12 is the equivalent of all extensions of the screen 12xx, excluding any explicitly listed longer codes. If a code of 123 is also listed, the 12 code is equivalent of all extensions of the screen 12xx except extensions of the screen 123x. The coding precludes having to list all the applicable 12xx extensions.
Valid Entry 0 to 16 blank
attd
Usage
Accepts up to 16 digits depending on the administered extension length. When 0 alone is entered, the administered extension length must be 1 and the DDD number must be 10 digits.
Generates a private calling number for a call from the attendant group.
Related links Ext Len on page 634
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Numbering-Private Format
Ext Len
Valid Entry 0 to 16 blank
Usage The number of digits the extension can have. Corresponds to the extension lengths allowed by the dial plan.
Maximum Entries
The maximum number of private numbering entries that can be administered on the system.
Private Prefix
The number that is added to the beginning of the extension to form a Private Identification Number. The length of the prefix and the extension must at least equal the total length.
Total Administered
The number of private numbering entries currently administered on the system.
Total Len
Valid Entry 0 to 16
Usage The total number of digits to send.
Trk Grp(s)
Communication Manager generates the station's identification number if Ext Code is administered, and this field is administered with the trunk group number carrying the call. Accepts one to seven digits. If blank, the identification numbers are independent on the trunk group carrying the call.
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Related links Ext Code on page 633
Trk Grp(s)
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Chapter 98: Numbering -- Public/Unknown Format
Use this screen to specify the required digits for the Calling Number Information Element (IE) and the Connected Number Information Element (IE), in addition to the QSIG Party number, for any extension in Public and/or Unknown Number Plans. Use this screen for Automatic Route Selection (ARS) public trunks as well as SIP trunks. This screen supports the ISDN Call Identification Display feature. This feature provides a name and number display for display-equipped stations within an ISDN network. The system displays the name and number of the caller on the telephone display of the called party and displays the name and the number of the called party on the telephone display of the caller. You must administer this screen if either Send Calling Number or Send Connected Number is specified, or Supplementary Service Protocol is b on the Trunk Group screen. You must administer the following in the public numbering table on the Numbering-Public/Unknown Format screen:
· Extensions of all calls traversing SIP trunks · Extensions terminating on SIP integrated Modular Messaging. However, do not configure a
prefix for the SIP trunk group by which the Modular Messaging is integrated. Note: If an entry applies to a SIP connection to Avaya Aura® Session Manager, the resulting number must be a complete E.164 number. Communication Manager automatically inserts the plus (+) digit. You can use the following command to modify the fields on this screen: change public-unknown-numbering n, where n is the extension length from 0 to 16. Related links Format on page 525 Send Calling Number on page 530 Send Connected Number on page 531 Supplementary Service Protocol on page 933
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CPN Prefix
CPN Prefix
Use this field to enter the number that is added to the beginning of the extension to form a Calling Party Number (CPN) number.
You must enter only digits. You cannot enter leading spaces, or spaces in between the digits. You can enter up to 15 digits. Communication Manager formulates the CPN number in the following manner:
· If the length of CPN Prefix matches Total CPN Length, the extension number is not used to formulate the CPN number.
· If the number of digits in CPN Prefix plus the extension length exceeds the administered Total CPN Length, Communication Manager deletes excess leading digits of the extension when formulating the CPN number.
· If the number of CPN Prefix digits plus the extension length is less than the Total CPN Length, Communication Manager does not allow the entry.
· If the Total CPN Length is 0, no calling party number information is provided to the called party and no connected party number information is provided to the calling party.
Note:
If you leave this field blank, the extension is sent unchanged.
Ext Code
In the case of a four-digit Ext Len, an Ext Code of 12 is the equivalent of all extensions of the screen 12xx, excluding any explicitly listed longer codes. If a code of 123 is also listed, the 12 code is equivalent of all extensions of the screen 12xx except extensions of the screen 123x. The coding precludes having to list all the applicable 12xx extensions.
Valid Entry 0 to 16 blank
attd
Usage
Accepts up to 16 digits depending on the administered extension length. When 0 alone is entered, the administered extension length must be 1 and the DDD number must be 10 digits.
Generates a private calling number for a call from the attendant group.
Related links Ext Len on page 634
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Numbering -- Public/Unknown Format
Ext Len
Valid Entry 0 to 16 blank
Usage The number of digits the extension can have. Corresponds to the extension lengths allowed by the dial plan.
Total CPN Len
Valid Entry 0 to 15 blank
Usage The number of digits the extension can have. Used when deleting an entry. This is the default.
Trk Grp(s)
Communication Manager generates the station's identification number if Ext Code is administered, and this field is administered with the trunk group number carrying the call. Accepts one to seven digits. If blank, the identification numbers are independent on the trunk group carrying the call.
Related links Ext Code on page 633
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Chapter 99: Off-PBX Telephone Mobile Feature Extensions
Example command: change off-pbx-telephone mobile-feature-ext
Mobile Call (CTI) Extension
A CTI call to this Mobile Feature Extension (MCE) creates an OPTIM call under CTI influence. A call to the MCE triggers an OPTIM extend-call from a desk phone to its mapped cell phone number and to the destination. All calls made using the MCE appear to the destination as if they were dialed from the desk phone.
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Chapter 100: Partition Routing Table
Identifies routing for partition groups associated with an ARS analysis entry. Example command: change partition-route-table n, where n is the routing index.
PGN 1 (through PGN 8)
The routing for each partition group associated with each route index number.
Valid Entry
1 to 2000 for all templates except 254 for Avaya Aura® Main Embedded template.
r1 to r32
node
deny
Usage The route pattern used to route the call.
The remote home numbering plan area table used to route the call. Node number routing. Blocks the call.
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Chapter 101: Personal CO Line Group
Sets up a personal central office line group. Example command: add personal-CO-line n, where n is the personal central office line group number.
Personal CO Line Group: page 1
CDR Reports
Valid Entry y
n r (ring-intvl)
Usage
All outgoing calls on this personal central office line group generate call detail records. If Record Outgoing Calls Only is disabled for CDR, incoming calls on this personal central office line group also generate call detail records.
Calls over this personal central office line group do not generate call detail records.
CDR records are generated for both incoming and outgoing calls. In addition, the following ringing interval CDR records are generated:
· Abandoned calls: The system creates a record with a condition code of "H" indicating the time until the call was abandoned.
· Answered calls: The system creates a record with a condition code of "G" indicating the interval from start of ring to answer.
· Calls to busy stations: The system creates a record with a condition code of " I " indicating a recorded interval of 0.
Related links Record Outgoing Calls Only on page 111
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Personal CO Line Group
Coverage Path
Valid Entry digits
t1 to t999 blank
Usage
The number of the call coverage path to use for incoming calls. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
The number of a time-of-day coverage table.
No coverage path is assigned. Assigning a coverage path is optional.
Data Restriction
Enables or disables data restriction that is used to prevent tones, such as call-waiting tones, from interrupting data calls. Data restriction provides permanent protection and cannot be changed by the telephone user. Cannot be assigned if Auto Answer is administered as all or acd. If enabled, whisper page to this station is denied.
Related links Auto Answer on page 765
Group Name
A unique name that provides information about the personal CO line group. Accepts up to 27 characters.
This field should contain names that identify the vendor and function of the personal CO line group rather than the group type (WATS).
Note:
Supported by Unicode language display for the 4610SW, 4620SW, 4621SW, and 4622SW, 1600-Series, 9600-Series IP Deskphones, 9600-Series H.323 Deskphones, and Avaya J100 Series IP Phones. Unicode is also an option for the 2420J telephone when the Display Character Set is katakana. For more information about 2420J, see 2420 Digital Telephone User's Guide.
Related links Display Character Set on page 821
Group Number
The personal CO line group number.
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Personal CO Line Group: page 1
Group Type
The type of personal central office line group. The fields that are displayed and available might change according to the personal central office line group type selected.
Valid Entry CO FX
WATS
Usage
Typically used to connect Communication Manager to the local telephone company central office, but can also connect adjuncts such as external paging systems and data modules.
A local telephone company central office (CO) trunk that connects the server running Communication Manager directly to a CO outside the local exchange area. Used to reduce long-distance charges if the organization averages a high volume of longdistance calls to a specific area code.
Used to reduce long-distance bills when your organization regularly places many calls to a specific geographical area in North America. With outgoing WATS service, calls can route to certain areas ("WATS band") for a flat monthly charge. With incoming WATS trunks, toll-free calling can be used to call customers and employees.
Outgoing Display
With Outgoing Display, telephones can show the name and number of the personal central office line group used for an outgoing call before the call is connected.
Valid Entry y n
Usage Displays the personal central office line group name and number. Displays the digits the caller dials.
Security Code
The station security code that users must dial to retrieve voice messages and to use the Demand Print Message feature. Accepts from three to eight digits.
TAC
The trunk access code (TAC) that must be dialed to access the personal central office line group. A different TAC must be assigned to each personal central office line group. CDR reports use the TAC to identify each personal central office line group. The characters "*" and "# " can be used as the first character in a TAC. Accepts a one- to four-digit number.
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Personal CO Line Group
TRUNK PARAMETERS
Analog Loss Group
Determines which administered two-party row in the loss plan applies to this personal central office line group if the call is carried over an analog signaling port in the personal central office line group.
Valid Entry 1 to 17
Usage
The index into the loss plan and tone plan. If values are administered other than in between 6 and 10 or 15 and 17, a warning message displays stating that the loss group might not be appropriate for this personal central office line group.
Answer Supervision Timeout
Valid Entry 0 to 250
Usage
The number of seconds Communication Manager waits before it acts as though answer supervision has been received from the far-end. During a cut-through operation, timing begins after each outgoing digit is sent and timing ceases after the far-end sends answer supervision. On senderized operation, the timer begins after the last digit collected is sent.
Note:
This field's setting does not override answer supervision sent from the network or from DS1 port circuit timers.
Related links Receive Answer Supervision on page 649 Administer Timers on page 516
Call Still Held
If enabled, the system prevents glare by extending the Incoming Glare Guard timer and delaying an outgoing seizure of a trunk for at least 140 seconds after it is released from an incoming call. This field is used when the receiving end media server or switch initiates the disconnection of incoming calls. This field affects only TN438B, TN465B, and TN2147 ports and is used primarily when the Country code is 2.
Available only for co or fx personal central office line groups.
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Related links Group Type on page 643
Personal CO Line Group: page 1
Charge Conversion
Available only for outgoing or two-way CO, FX, and WATS trunk groups.
Valid Entry 1 to 64 500
Usage
Communication Manager multiplies the number of charge units by the value of this field and displays it as a currency amount. Without a value in this field, Communication Manager displays the number of charge units without converting it to currency.
Related links Direction on page 507 Trunk Direction on page 649
Charge Type
Text string used to describe charges related to a telephone call. The system displays these words or characters on telephone screen after the charge amount. Typically uses either the currency symbol or the charge type, but not both. Accepts up to seven characters. Embedded spaces count as characters.
Available only for outgoing or two-way CO, FX, and WATS personal central office line groups.
Related links Direction on page 507
Country
The country code that corresponds to the protocol used by the local telephone company central office (CO) where the personal central office line group terminates.
Available only for personal central office line groups that connect Communication Manager to a CO in the public network -- CO, FX, and WATS personal central office line groups.
Valid Entry
1 to 25, except for 19
Usage For a list of country codes, see the Country code table.
Related links Country options table on page 817 Trunk Gain on page 649
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Personal CO Line Group Trunk Termination on page 651
Currency Symbol
The telephone displays the currency symbol before the charge amount. Accepts from one to three characters. Leading and embedded spaces count as characters.
Available only for outgoing or two-way CO, FX, and WATS personal central office line groups.
Related links Trunk Direction on page 649 Direction on page 507
Decimal Point
The appropriate representation for a decimal point as seen on the telephone displays. Available only with outgoing or two-way CO, FX, and WATS personal central office line groups. Also available for isdn personal central office line groups, if the Charge Advice field is set to a value other than none.
Note:
If the received charge contains no decimals, no decimal point is displayed (that is, the administered decimal point is ignored for charge information received with no decimals). On a QSIG personal central office line group, unlike other personal central office line groups, the Decimal Point field is independent of whether the calling screen displays a decimal point. Instead, it tells what symbol should be displayed if the QSIG AOC received has a 1/10 or 1/100 or 1/1000 Multiplier.
Valid Entry comma
period
none
Usage
If the received charge contains decimals, the charge is displayed at the calling endpoint's display with a comma as the decimal point. Divides the charge value by 100.
This is the default. If the received charge contains decimals, the charge is displayed at the calling endpoint's display with a period as the decimal point. Divides the charge value by 100.
No decimal point is displayed.
Related links Charge Advice on page 506 Direction on page 507 Trunk Direction on page 649
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Personal CO Line Group: page 1
Digital Loss Group
Valid Entry 1 to 19
Usage
Determines which administered two-party row in the loss plan applies to this personal central office line group if the call is carried over a digital signaling port in the personal central office line group. If values other than 18 or between 11 and 15 are administered, a warning message displays stating that the loss group is inappropriate for this personal central office line group.
Disconnect Supervision-In
Indicates whether Communication Manager receives disconnect supervision for incoming calls over this personal central office line group.
Available only for incoming or two-way personal central office line groups.
Valid Entry y
n
Usage
Allows trunk-to-trunk transfers involving trunks in this group. The far-end server or switch sends a release signal when the calling party releases an incoming call, and the far-end server or switch is responsible for releasing the trunk. Enhances Network Call Redirection.
The far-end server or switch does not provide a release signal, the hardware cannot recognize a release signal, or timers are preferred for disconnect supervision on incoming calls. Prevents trunk-to-trunk transfers involving trunks in this group.
Caution:
In general, U.S. local telephone company central offices provide disconnect supervision for incoming calls but not for outgoing calls. Public networks in most other countries do not provide disconnect supervision for incoming or outgoing calls. Check with the network services provider.
Related links Direction on page 507 Trunk Direction on page 649
Outgoing Dial Type
Sets the method used to transmit digits for an outgoing call. Usually, this method should match what the local telephone company central office provides.
Available for CO, FX, and WATS personal central office line groups. This field is also available for the trunks with the group type as Tie and when the Trunk Signaling Type is blank, cont, or dis.
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Personal CO Line Group
Valid Entry tone rotary automatic mf
r1mf mf2/6
Usage
Uses Dual Tone Multifrequency (DTMF) addressing, also known as "touchtone" in the U.S. With this, the personal central office line group can support both DTMF and rotary signals. For pulsed and continuous E&M signaling in Brazil and for discontinuous E&M signaling in Hungary, use tone or mf.
With this, only the dial pulse addressing method can be used by non-touch tone telephones. For example, this value is appropriate for an internal full touch tone system and for a connection to a local telephone company central office that only supports rotary dialing.
For tie trunks if the Trunk Signaling Type is not administered. This provides "cutthrough" operation to outgoing callers who dial a trunk access code, connecting them directly to local telephone company central office dial tone and bypassing any toll restrictions administered on Communication Manager.
Use this option if the Trunk Signaling Type field is blank. The system displays this field only if you set the Multifrequency Signaling field on the System Parameters Customer-Options screen to y. You cannot use this option if the Used for DCS field is y. For pulsed and continuous E&M signaling in Brazil and for discontinuous E&M signaling in Hungary, use tone or mf.
Enter r1mf for CAMA trunk groups. It is the only outgoing dial type allowed on CAMA trunk groups. Enter r1mf to allow Russian MF Packet Signaling on outgoing trunks. Russian MF Packet Signaling carries calling party number and dialed number information. Group type field must be set to co.
Use this entry for signaling in Spain. This signaling type provides normal public network service and a virtual private network digital trunk service called IBERCOM. Use this entry only if the Trunk Signaling Type field is blank.
The system displays this entry only if you enable Multifrequency Signaling for the system. The system does not display this entry if the trunk is used for DCS.
Related links Group Type on page 508 Multifrequency Signaling on page 837 Trunk Signaling Type on page 917
Prefix-1
If enabled, the prefix "1" is added to the beginning of the digit string for outgoing calls. Use this field for outgoing and two-way personal central office line groups handling long distance service. Disable for personal central office line groups in AAR or ARS route patterns.
Available only for CO and FX personal central office line groups.
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Personal CO Line Group: page 1
Receive Answer Supervision
If enabled, the Communication Manager provides answer supervision for a trunk group. For Outbound Call Management applications, use for trunks supporting network answer supervision. For trunks that do not receive a real answer, this field determines when the CallVisor AdjunctSwitch Application Interface (ASAI) connect event is sent.
Related links Answer Supervision Timeout on page 517 Administer Timers on page 516
Trunk Direction
Important: This setting must match the provider's settings. The direction of the traffic on this personal central office line group.
Valid Entry incoming outgoing two-way
Usage
Traffic on this personal central office line group is incoming.
Traffic on this personal central office line group is outgoing.
Traffic on this personal central office line group is incoming and outgoing. Required for Network Call Redirection.
Related links Charge Conversion on page 524
Trunk Gain
Specifies the amplification applied to the trunks in this group. With the values administered for Trunk Termination and Country code, the value in this field also determines the input and transhybrid balance impedance for TN465B, TN2146, TN2147, and TN2184 ports. All other CO circuit packs are set automatically to high.
Valid Entry high low
Usage Used if users complain of low volume. Used if users complain of squeal or feedback.
Related links Trunk Termination on page 651 Country on page 645
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Personal CO Line Group
Trunk Port
The port to which the trunk is connected.
Trunk Type
Controls the seizure and start-dial signaling used on this personal central office line group. Entries in this field vary according to the function of the personal central office line group and must match the corresponding setting on the far-end server or switch.
Available only for CO, FX, and WATS personal central office line groups.
Valid Entry ground-start loop-start auto/auto auto/delay auto/immed auto/wink
2-wire-ac 2-wire-dc 3-wire
Usage
Use ground-start signaling for two-way trunks whenever possible. Ground-start signaling avoids glare and provides answer supervision from the far end.
In general, loop-start signaling is used only for one-way trunks. Loop-start signaling is susceptible to glare and does not provide answer supervision.
The term before the slash tells Communication Manager how and when it receives incoming digits. The term after the slash tells Communication Manager how and when it should send outgoing digits.
· auto -- Used for immediate connection to a single preset destination (incoming central office trunks, for example). No digits are sent, because all calls terminate at the same place.
· delay -- The sending server running Communication Manager does not send digits until it receives a delay dial signal (an off-hook signal followed by an on-hook signal) from the far-end server or switch, indicating that it is ready to receive the digits.
· immed -- The sending server running Communication Manager sends digits without waiting for a signal from the far-end server or switch.
· wink -- The sending server running Communication Manager does not send digits until it receives a wink start (momentary off-hook) signal from the far-end server or switch, indicating that it is ready to receive the digits.
These entries are used with local telephone company central office (CO) trunks in Russia. The specific CO should match one of these values.
Available only if the Country code is 15 and the CO trunks use ports on a TN2199 circuit board.
Related links Country on page 645
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Personal CO Line Group: page 2
Trunk Termination
Adjusts the impedance of the personal central office line group for optimal transmission quality.
Valid Entry 600ohm
rc
Usage
The distance to the local telephone company central office (CO) or to the server at the other end of the trunk is less than 3,000 feet.
The distance to the CO or to the server at the other end of the trunk is more than 3,000 feet.
Personal CO Line Group: page 2
Ext
The extension of telephones that have a CO Line button.
Name
The name assigned to telephones that have a CO Line button.
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Chapter 102: Pickup Group
Implements call pickup groups. A pickup group is a group of users authorized to answer calls to a telephone extension within that group of users. A telephone extension can belong to only one pickup group, and a pickup group can have up to 50 extensions. Example command: add pickup-group n, where n is the pickup group number.
GROUP MEMBER ASSIGNMENTS
Ext
The extension of the pickup group number. A VDN cannot be assigned to a Call Pickup group.
Name
The name assigned to the pickup group number extension.
Extended Group Number
Available only if the Extended Group Call Pickup is administered as flexible.
Valid Entry 1 to 100 blank
Usage
The Extended Group number. The extended group is a collection of pickup groups that can answer calls to a telephone extension within that other pickup group.
Related links Extended Group Call Pickup on page 297
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Group Number
The pickup group number.
Group Number
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Chapter 103: Policy Routing Table
With Policy Routing Table, you can distribute calls among a set of call centers based on specified percent allocation. Various types of incoming calls that arrive at a particular VDN can be directed to a Policy Routing Table (PRT) instead of to a vector. The PRT then distributes the calls to the administered Route-to VDNs based on the specified percent allocation targets. Use this screen to implement and monitor percentage allocation routing by assigning destination routes and target percentages. Example command: change policy-routing-table n, where n is the policy routing table number.
Actual %
The actual percent of total calls routed to a VDN. Calculated to six decimal places, but only the first decimal place is displayed.
Call Counts
The current number of calls routed to a VDN.
Index
The sequential number of the row in a Policy Routing Table.
Name
The name of the Policy Routing Table (PRT). Accepts a string of up to 15 alphanumeric characters.
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Number
The number of the Policy Routing Table (PRT).
Number
Period
The period for resetting the call counts and actual percentages.
Valid Entry 100_count
max_count
Half-hour hour daily weekly
Usage
Resets the call counts (and displayed %) when total calls for the PRT reach 100. At this point, the total calls match the target routing pattern percentages. This ensures that the routing points have equal distribution of calls all the time. This is the default.
Call counts are maintained until calls delivered to at least one of the VDNs exceed 65,400. At this point, calls continue to be distributed over the VDNs but the call counts are reset when the actual percentages equal the targets for all of the VDNs at the same time.
Resets the call counts at the top of the hour and at the 30minute point.
Resets the call counts at the top of the hour.
Resets the call counts at midnight, every night.
Resets the call counts at midnight on Saturday.
Route-to VDN
VDN extension to which calls are to be routed. Accepts up to 15 extensions containing 1 to 16 digits.
Target %
Valid Entry 0 to 100
Usage
The target percent of total calls to be routed to a VDN. Use whole numbers only, no decimal fractions.
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Policy Routing Table
Totals
Displays totals for Target % and Call Counts for all the assigned VDNs in the PRT. The total for Target % is always 100 for form submittal.
Type
The type of algorithm the Policy Routing Table (PRT) supports.
VDN Name
The name of a VDN, if a name has been assigned previously.
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Chapter 104: Precedence Routing Digit Analysis Table
Communication Manager compares dialed numbers with the dialed strings in this table and determines the route pattern of an outgoing Multiple Level Precedence and Preemption (MLPP) call. Example command: change precedence-routing analysis n, where n is the precedencerouting digit analysis table number.
Destination Code Control
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
Use Destination Code Control (DCC) to restrict outbound calls to a specific called number or a range of numbers to keep the trunk free for important calls.
DCC limits outbound calls from the outgoing budget (IPBo).
To enable the DCC feature you must first activate DCC by using FAC in COS.
Valid entry y n
Usage DCC is activated. DCC is not activated. The default value is n.
Dialed String
Communication Manager matches the dialed numbers with the entry in the Dialed String field that most closely matches the dialed number. You can enter up to 18 digits that the call-processing server analyzes. You can also enter the wildcard characters, x and X.
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Precedence Routing Digit Analysis Table
Max
Use this field to specify the maximum number of user-dialed digits that the system must collect to match the dialed string.
Min
Use this field to specify the minimum number of user-dialed digits that the system must collect to match the dialed string.
Percent Full
This field displays the percentage of system memory resources used for the AAR and the ARS digit analysis. For more information about the system memory resources, see Avaya Aura® Communication Manager System Capacities Table.
Percentage IPBO Blocked
Note: This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers. With Percentage IPBO Blocked, Communication Manager supports blocking a percentage of calls to a destination for the DCC feature. You can enter a value from 1% to 100%. The default value is 100%.
Preempt Method
The preemption method used by the server running Communication Manager for this dialed string.
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Valid Entry group
route
Preempt Method
Usage The system checks the first trunk group in the route pattern to determine if any trunks are idle. If the system finds an idle trunk, the system connects the call. This is the default. The system checks each trunk group in the route pattern to determine if any trunks are idle. If the system finds an idle trunk, the call is connected.
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Chapter 105: Precedence Routing Digit Conversion Table
Assigns the Precedence Routing digit conversion. Digit conversion takes digits dialed on incoming calls and converts the digits to local telephone numbers, usually extension numbers. Example command: change precedence-routing digit-conversion n, where n is the precedence routing digit conversion table number.
Conv
Use this field to enable additional digit conversion.
Del
Use this field to enter the number of digits that the system must delete from the starting of the dialed string.
Matching Pattern
Use this field to specify the pattern that the system must use to match the dialed number. The pattern can be up to 18 characters long and consist of digits and the wildcard characters, x and X.
Max
Use this field to specify the maximum number of user-dialed digits that the system must collect to match the dialed string.
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Min
Min
Use this field to specify the minimum number of user-dialed digits that the system must collect to match the dialed string.
Net
Valid Entry ext
pre
Usage
Communication Manager routes the call as an extension using the Dial Plan Analysis and Uniform Dial Plan forms, as appropriate.
Precedence routing. Uses the Precedence Analysis Tables to route the call.
Replacement String
Use this field to specify the digit string that must replace the deleted portion of the dialed number.
Valid entry blank 0 to 9, *
#
Usage To delete the digits without replacement. To type the digit string. The digit string can have up to 18 characters. To indicate the end of dialing.
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Chapter 106: PRI Endpoint
Administers PRI Endpoints for the Wideband Switching feature.
Note: A PRI Endpoint with a width greater than 1 can be administered only if Wideband Switching has been enabled for the system. A PRI Endpoint is an endpoint application connected to line-side ISDN-PRI facilities and has standard ISDN-PRI signaling interfaces to the system. A PRI Endpoint is defined as 1 to 31 adjacent DS0s/B-channels, addressable via a single extension, and signaled via a D-channel (Signaling Group) over a standard T1 or E1 ISDN-PRI interface. Example command: add pri-endpoint n, where n is the extension number. Related links Wideband Switching on page 842
COR
Valid Entry 0 to 995
Usage The class of restriction (COR) used to determine calling and called party privileges.
COS
Valid Entry 0 to 15
Usage
The Class of Service (COS) used to determine the features that can be activated by, or on behalf of, the endpoint.
Extension
The extension number used to access the PRI endpoint.
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Maintenance Tests
If enabled, runs hourly maintenance tests on this PRI Endpoint.
Maintenance Tests
Name
The name of the endpoint. Accepts up to 27 alphanumeric characters.
Originating Auto Restoration
If enabled, automatically restores calls originating from this PRI Endpoint (while maintaining endpoint call status) in the case of network failure if the call is over SDDN network facilities.
Signaling Group
Valid Entry digits or blank
Usage
The D-channel or D-channel pair that provides the signaling information for the set of B-channels that make up the PRI Endpoint. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Simultaneous Calls
If enabled, specifies that multiple simultaneous calls can be placed to/from the PRI Endpoint.
(Starting) Port
The seven-character starting port of the PRI Endpoint.
Valid Entry 1 to 64 A to E
Usage First and second characters are the cabinet number. Third character is the carrier.
Table continues...
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PRI Endpoint
Valid Entry 0 to 20 01 to 04 (Analog TIE trunks)
01 to 31
Usage Fourth and fifth character are the slot number. Six and seventh characters are the circuit number.
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
Width
Valid Entry 1 to 31
Usage
The number of adjacent DS0 ports beginning with the administered Starting Port, that make up the PRI Endpoint. This field cannot be blank. A width of 6 defines a PRI Endpoint that can support data rates up to 384 Kbps.
Related links (Starting) Port on page 663
WIDEBAND SUPPORT OPTIONS
Contiguous
Available only if "N by DS-zero" (NXDS0) multi-rate service is enabled.
Valid Entry y
n
Usage
Specifies the "floating" scheme. NXDS0 calls are placed on a contiguous group of Bchannels large enough to satisfy the requested bandwidth without constraint on the starting channel (no fixed starting point trunk). Unavailable with H0 ISDN information transfer rate.
Specifies the "flexible" scheme. NXDS0 calls are placed on any set of B-channels on the same facility as long as the requested bandwidth is satisfied. There are no constraints, such as contiguity of B-channels or fixed starting points
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Related links NxDS0 on page 545 NXDS0 on page 665
WIDEBAND SUPPORT OPTIONS
H0
If enabled, specifies the ISDN information transfer rate for 384 Kbps of data, which is comprised of six B-channels. When a PRI Endpoint is administered to support H0, the hunt algorithm to satisfy a call requiring 384 Kbps of bandwidth uses a fixed allocation scheme.
H11
If enabled, specifies the ISDN information transfer rate for 1536 Kbps of data, which is comprised of 24 B-channels. When a PRI Endpoint is administered to support H11, the hunt algorithm to satisfy a call requiring 1536 Kbps of bandwidth uses a fixed allocation scheme.
H12
If enabled, specifies the ISDN information transfer rate for 1920 Kbps data, which includes 30 Bchannels. When a PE is administered to support H12, the hunt algorithm to satisfy a call requiring 1920 Kbps of bandwidth uses a fixed allocation scheme.
NXDS0
If enabled, specifies the NXDS0 multi-rate service.
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Chapter 107: Processor Channel Assignment
Assigns each local processor channel to an interface link channel, and defines the information associated with each processor channel on an Ethernet link.
Note: You cannot remove a service from this screen if that service has overrides defined on the Survivable Processor screen. Example command: change communication-interface processor-channels
Appl
Specifies the server application type or adjunct connection used on this channel.
Valid Entry audix ccr dcs fp-mwi
gateway gateway-tcp mis qsig-mwi
Usage
Voice Messaging.
Contact Center Reporting, now known as Avaya IQ.
Distributed Communication System.
ISDN Feature Plus Message Waiting Indication. This channel passes message waiting light information for subscribers on the messaging system, from a messaging adjunct on a main switch for a telephone on a satellite switch. The terminating location (far end) of this channel must be a Communication Manager system compatible with ISDN Feature Plus proprietary protocol.
Supports an X.25 connected AUDIX connected to an ISDN DCS network.
Supports a TCP-connected voice messaging system connected to an ISDN DCS network.
Management Information System, otherwise known as the Call Management System.
QSIG Message Waiting Indication. Used with a QSIG-based interface to a messaging system, this channel passes message waiting light information for subscribers on the messaging system.
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Destination Node
Destination Node
Identifies the server or adjunct at the far end of this link. The destination node can be an adjunct name, server name, far end IP address, or node name for services local to the Avaya server. For ppp connections, match the Destination Node Name administered for the ppp data module.
Destination Port
Valid Entry 5000 to 64500 0
Usage The number of the destination port. Any port can be used.
Enable
Enables or disables this processor channel on the main server.
Gtwy to
A number that identifies the processor channel to which the specified processor channel is serving as a gateway.
Interface Channel
The channel number or the TCP/IP listen port channel to carry this processor (virtual) channel.
Valid Entry 5000 to 64500
0
Usage
For ethernet or ppp. For TCP/IP, interface channel numbers are in the range 5000 to 64500. The value 5001 is recommended for CMS, and 5003 is recommended for DCS.
Any port can be used.
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Processor Channel Assignment
Interface Link
Valid Entry 1 to 254 pv4 or pv6
blank
Usage
The physical link carrying this processor (virtual) channel.
Communication Manager's Processor Ethernet interface used for adjunct connectivity.
Not administered.
Mach ID
Valid Entry 1 to 63 for MWI
1 to 63 for DCS
1 to 99 for voice messaging
blank
Usage The destination server ID defined on the dial plan of the destination server.
Related links Mach ID on page 727
Mode
Valid Entry c(lient) s(erver) blank
Usage Indicates whether the IP session is passive (client) or active (server).
This field must be blank if the interface link is procr-intf. This field cannot be blank if the type of interface link is ethernet or ppp.
Proc Chan
The number assigned to each processor channel.
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Session - Local/Remote
Session - Local/Remote
Valid Entry 1 to 384 blank
Usage
The Local and Remote Session numbers. For each connection, the Local Session number on the Avaya server must equal the Remote Session number on the remote server and vice versa. It is allowed, and sometimes convenient, to use the same number for the Local and Remote Session numbers for two or more connections. Local and Remote Session numbers must be consistent between endpoints.
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Chapter 108: QSIG to DCS TSC Gateway
This screen determines when and how to convert messages from a QSIG NCA-TSC to an administered AUDIX NCA-TSC. This screen maps the QSIG subscriber number to the appropriate AUDIX signaling group and TSC index. Available only if Interworking with DCS is enabled for the system. Example command: change isdn qsig-dcs-tsc-gateway Related links
Interworking with DCS on page 849
Sig Grp
Valid Entry digits
Usage
The assigned signaling group number. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Subscriber Number
A subscriber number up to 20 characters in length. Accepts *, x, and X as wildcard characters.
Note:
If the Subscriber Number is local, the form does not send any DCS messages over the listed Sig Grp/TSC Index. Therefore, you can put a block entry like '2xxx' with the understanding that if extension 2001 is local, no DCS message will be sent.
TSC Index
The TSC Index for each machine ID.
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Valid Entry 1 to 64
TSC Index
Usage The assigned signaling group number for the qsig-mwi application type.
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Chapter 109: Reason Code Names
Assigns names to reason codes. Each reason code can have a different name for Aux Work and for Logout. The system displays the screen only when you set the Two-Digit Aux Work Reason Codes to y.
Note: Logout reason codes can only be in the range of 0 to 9, even if Two-Digit Aux Work Reason Codes is active. Example command: change reason-code-names
Aux Work
The name associated with a reason code when the agent uses the reason code to enter Aux Work mode. Accepts up to 16 alphanumeric characters.
Default Reason Code
The default reason code names. A separate name can be administered for the Aux Work Reason Code of 0 and for the Logout Reason Code of 0. If an agent changes to Aux Work mode and the Aux Work Reason Code Type is set to none, the agent is put into Aux Work mode with the default Aux Work reason code, even if a different reason code is administered for the Aux button. If an agent logs out when the Logout Reason Code Type is set to none, the agent is logged out with the default Logout reason code. Accepts up to 16 alphanumeric characters.
Interruptible
Specifies whether or not each reason code is interruptible.
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Logout
Note: The Default Reason Code, Auto-answer IP Failure Aux Work Reason Code, and Maximum Agent Occupancy Aux Work Reason Code cannot be made interruptible.
Logout
The name associated with a reason code when the agent uses the reason code to log out. Accepts up to 16 alphanumeric characters.
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Chapter 110: Remote Access
Implements the Remote Access feature. Remote Access permits a caller located outside the system to access the system through the public or private network and then use the features and services of the system.
Security alert: The Remote Access feature, when properly administered, enables the customer to minimize the ability of unauthorized persons to gain access to the network. It is the customer's responsibility to take the appropriate steps to properly implement the features, evaluate and administer the various restriction levels, protect access codes, and distribute them only to individuals who have been advised of the sensitive nature of the access information. Each authorized user should be instructed concerning the proper use and handling of access codes. In rare instances, unauthorized individuals make connections to the telecommunications network through use of remote access features. In such an event, applicable tariffs require the customer pay all network charges for traffic. Avaya will not be responsible for such charges, and will not make any allowance or give any credit for charges that result from unauthorized access. Example command: change remote-access
Authorization Code Required
If enabled, requires an authorization code be dialed by Remote Access users to access the system's Remote Access facilities. The use of an authorization code in conjunction with a barrier code increases the security of the Remote Access feature.
Barrier Code
The number users must dial to use Remote Access. Can be used with an authorization code. Accepts a four- to seven-digit number in any combination of digits. Must conform to the administered Barrier Code Length. The none value is required if a Barrier Code Length is not administered. Related links Barrier Code Length on page 675
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Barrier Code Length
Barrier Code Length
Valid Entry 4 to 7 blank
Usage
The length of the barrier code. A barrier code length of 7 provides maximum security.
Calls Used
The number of calls placed using the corresponding barrier code. A usage that exceeds the expected rate indicates improper use.
COR
Valid Entry 0 to 995
Usage
The number of the class of restriction (COR) associated with the barrier code. To provide maximum security, assign the most restrictive COR that provides only the level of service required.
COS
Valid Entry 0 to 15
Usage
The number of the class of service (COS) associated with the barrier code. To provide maximum security, assign the most restrictive COS that provides only the level of service required.
Disable Following a Security Violation
Available only if SVN Authorization Code Violation Notification is enabled for the system.
Valid Entry y
Usage
Disables the Remote Access feature following detection of a Remote Access security violation. The system administrator can re-enable Remote Access using the enable remote-access command.
Table continues...
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Remote Access
Valid Entry n
Usage
Enables the Remote Access feature following detection of a Remote Access security violation.
Related links SVN Authorization Code Violation Notification Enabled on page 694
Expiration Date
The date the barrier code expires. Assign an expiration date based on the expected length of time the barrier code is needed. For example, if the barrier code is expected to be used for a two-week period, assign a date two weeks from the current date. If both an Expiration Date and No. of Calls are assigned, the corresponding barrier code expires when the first of these criteria is satisfied.
No. of Calls
Valid Entry 1 to 9999 blank
Usage
The number of Remote Access calls that can be placed using the associated barrier code. If both an Expiration Date and No. of Calls are assigned, the corresponding barrier code expires when the first of these criteria is satisfied.
Related links Expiration Date on page 676
Permanently Disable
If enabled, permanently blocks remote access to the administration interface. Reactivation of remote access to the interface requires the intervention of Avaya Services.
Remote Access Dial Tone
Enables or disables the user hearing dial tone as a prompt for entering the authorization code. Disabling this feature provides maximum security. Available only if Authorization Codes are required for Remote Access.
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Related links Authorization Code Required on page 674
Remote Access Extension
Remote Access Extension
The extension assigned to handle Remote Access calls. The remote access extension is used as if it was a DID extension. Only one DID extension can be assigned as the remote access extension. Calls to that number are treated the same as calls on the remote access trunk.
When a trunk group is dedicated to Remote Access, the remote access extension number is administered as the trunk group's incoming destination.
A Vector Directory Number (VDN) extension cannot be used as the remote access extension.
Can be blank if no barrier codes are administered.
Related links Incoming Destination on page 910
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
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Chapter 111: Remote Call Coverage Table
Provides automatic redirection of certain calls to alternate non-local answering positions in a coverage path. Non-local numbers can be any ARS or AAR number, any number on the public network, any international number, or a UDP/DCS extension up to 16 digits or blank, which includes any ARS/AAR facility access code, any trunk dial access code (TAC), long distance dialing code, or international dial code. Example command: change coverage remote n, where n is the remote coverage number.
011000
The identifier for the destination coverage point. Accepts up to 16 digits. Two places are used for L, D, %, and ,.
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Valid Entry * # L
D % , blank
011000
Usage DTMF digit asterisk DTMF digit pound Uses a coverage point only when in Survivable Remote or Survivable Core Server mode Represents the called extension digits Remaining digits are for end-to end signaling Pauses for 1.5 seconds Not administered
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Chapter 112: Remote Office
Supports the Remote Office feature, an arrangement whereby a user can set up a remote office without having an on-premises physical desk-set. An R300 is issued to connect remote DCP and analog telephones, IP telephones, and H.323 trunks to the Communication Manager server using IP. Example command: change remote-office n, where n is the assigned remote office number.
Location
Valid Entry 1 to 2000
blank
Usage
(Depending on your server configuration, see Avaya Aura® Communication Manager System Capacities Table.) Assigns the location number to the remote office comprised of the associated time zone and the appropriate numbering plan. See the Location sections in Avaya Aura® Communication Manager Feature Description and Implementation, for the other ways to assign a location, and for a list of features that use location.
Obtains the location from the cabinet containing the CLAN or the gateway that the endpoint registered with. By default, the value is blank.
Network Region
Valid Entry 1 to 2000 blank
Usage The network region assigned to all stations supported at this remote office.
Node Name
The node name of the remote office.
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Site Data
Any required site information. Accepts up to 30 alphanumeric characters.
Site Data
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Chapter 113: RHNPA Table
Defines route patterns for specific three-digit codes, usually direct distance dialing (DDD) prefix numbers. Example command: change rhnpa n, where n is the prefix number.
CODES
The 100-block of codes being administered.
Code-Pattern Choice Assignments
Valid Entry 1 to 24
Usage A pattern choice number associated with each office code.
Pattern Choices
Valid Entry 1 to 999 1 to 2000 blank
Usage
The maximum number of route patterns that you can configure depends on your system configuration. The route pattern number associated with each code. If you use one pattern for most of the codes, assign that pattern to choice 1.
RHNPA TABLE
The RHNPA table number.
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Chapter 114: Route Pattern
Use this screen to define route patterns. Each route pattern contains a list of trunk groups that Communication Manager uses to route a call. The maximum number of route patterns and trunk groups allowed depends on the system configuration and the memory available in the system. Use this screen to convert the dialed number into a format that the network uses for routing the call. For example, you can:
· Insert or delete digits when Automatic Alternate Routing (AAR) or Automatic Route Selection (ARS) calls route over different trunk groups.
· Convert an AAR number into an international number. · Insert an area code into an AAR number to convert an on-network number to a public network
number. · Insert the dial access code for an alternative carrier. The system uses this alternative carrier
when a call directly accesses a local telephone company central office (CO) and if the long distance carrier that the CO provides is unavailable. Network regions are added to the Route Pattern form. In the Route Pattern form, you can specify the network region that can be used against each route preference. Network region is applicable for SIP trunks and Off-PBX Telephone Integration and Mobility (OPTIM) end points like EC500 and Telecommuter. A network region can be populated for each of the route patterns. This network region will be used only when routing SIP trunk calls (SRE Calls) over the route preference (Trunk Group). Network region field on the route pattern applies only for trunk and OPTIM such as EC500 and telecommuter calls but not for SIP station calls. This overrides the network region field on the SIP signaling group for the above mentioned applicable calls. To modify the value of the fields on the Route Pattern screen, use the following command: change route-pattern n, where n is the route pattern number.
Band
A number that represents the OUTWATS band number (US only). Available only if Services/Features is administered as outwats-bnd and ISDN-PRI or ISDN-BRI Trunks are enabled for the system. Band is required by Call-by-Call Service Selection. Related links ISDN-BRI Trunks on page 835
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Route Pattern
ISDN-PRI on page 836 Service/Feature on page 691
BCC Value
Identifies the type of call appropriate for a trunk group. Available only if ISDN-PRI or ISDN-BRI Trunks are enabled for the system.
Valid Entry y/n
Usage
If enabled, (0, 1, 2, 3, 4, or W) indicates the BCC value is valid for the routing preference. A trunk group preference can have more than one BCC.
BCC values:
BCC Value 0 1 2 M 4 W
Description Voice-Grade Data and Voice 56-kbps Data (Mode 1) 64-kbps Data (Mode 2) Multimedia call 64-kbps Data (Mode 0) 128 to 1984-kbps Data (Wideband)
Related links ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
BCIE (Bearer Capability Information Element)
Determines how to create the ITC codepoint in the setup message. Applies to ISDN trunks. Available only if the Information Transfer Capability (ITC) is administered as both.
Valid Entry ept unr
Usage endpoint unrestricted
Related links ITC (Information Transfer Capability) on page 687
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CA-TSC Request
CA-TSC Request
For the DCS+ features, Communication Manager sends a Call-Associated Temporary Signaling Connection (CA-TSC) request to send messages other than normal call control messages.
Valid entry as-needed at-setup
none
Usage
Communication Manager sends the CA-TSC request only when needed.
Communication Manager sends the CA-TSC request for every B-channel call, whether or not it is needed.
Communication Manager does not send the CA-TSC request Use this entry to allow Communication Manager to send Non-Call-Associated Temporary Signaling Connection (NCA-TSC) requests.
DCS/QSIG Intw
Enables or disables DCS/QSIG Voice Mail Interworking. Available only if Interworking with DCS is enabled for the system.
Related links Interworking with DCS on page 849
FRL
Valid Entry 0 to 7
Usage
The Facility Restriction Level (FRL) associated with the group routing preference. 0 is the least restrictive, and 7 is the most restrictive. The calling party's FRL must be greater than or equal to this FRL to access the associated trunk group.
Security alert: For system security reasons, use the most restrictive FRL possible.
Grp No
Valid Entry 1 to 2000
Usage The trunk group number associated with the routing preference.
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Route Pattern
Hop Lmt
The number of hops for each routing preference. Communication Manager blocks a hop equal to or greater than the number entered.
Valid Entry blank 1 to 9 1 to 32
Usage There is no limit to the number of hops for this preference. Valid for limiting the number of hops if using the tandem hop feature. Valid for limiting the number of hops if using the transit feature.
Inserted Digits
The digits to insert for routing. Communication Manager can send up to 52 digits. This includes up to 36 digits entered here plus up to 18 digits originally dialed. Special symbols count as two digits each.
Valid Entry *
# , (comma)
+
% ! & p
Usage
When * is in the route pattern and the outgoing trunk is signaling type "mf", the MFC tone for the "end-of-digits" is sent out to the local telephone company central office (CO) in place of the *.
When # is in the route pattern and the outgoing trunk is signaling type "mf", the MFC tone for the "end-of-digits" is sent out to the CO in place of the #.
Creates a 1.5 second pause between digits being sent. Do not use as the first character in the string unless absolutely necessary. Misuse can result in some calls, such as Abbreviated Dialing or Last Number Dialed, not completing.
Wait for dial tone up to the Off Premises Tone Detection Timer and then send digits or intercept tone depending on system-wide feature administration. The length of waiting time is controlled by the Off-premises Tone Detect Timeout Interval field on the Feature-related system parameters screen.
Start End-to-End Signaling.
Wait for dial tone without timeout and then send DTMF digits.
Wait for ANI that is used for Russian pulse trunks.
The associated trunk group must be of type sip. The single digit p is used for fully qualified E.164 numbers. Communication Manager inserts a plus (+) digit to the digit string.
Related links Off-Premises Tone Detect Timeout Interval (seconds) on page 288 Outpulse Without Tone on page 326
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ITC (Information Transfer Capability)
ITC (Information Transfer Capability)
Identifies the type of data transmission or traffic that this routing preference can carry. The ITC applies only to data calls (BCC 1 through 4).
This field must be both or unre for a BCC Value of W.
Valid Entry both rest(ricted) unre(stricted)
Usage Calls from restricted and unrestricted endpoints can access the route pattern. Calls from restricted endpoints can access the route pattern. Calls from unrestricted endpoints can access the route pattern.
Related links BCC Value on page 684
IXC
Identifies the carrier, such as AT&T, used for calls that route using an Inter-Exchange Carrier (IXC), and for Call Detail Recording (CDR).
Available only if ISDN-PRI or ISDN-BRI Trunks are enabled for the system.
Valid Entry Valid carrier code user none
Usage
Identifies the carrier for IXC calls.
For presubscribed carrier. Used when an IXC is not specified.
Must be none for non-ISDN trunk groups and for Telcordia Technologies NI-2 Operator Service Access. If it is necessary to send an IXC code for a non-ISDN trunk group, the IXC code can be entered as Inserted Digits.
Related links Inserted Digits on page 686 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
LAR
The routing-preference for Look Ahead Routing.
Valid Entry next
Usage Go to the next routing preference and attempt the call again.
Table continues...
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Route Pattern
Valid Entry rehu none
Usage
Re-hunt within the current routing preference for another trunk to attempt the call again.
Look Ahead Routing is disabled for the preference.
No. Del. Digits
Modifies the dialed number so an AAR or ARS call routes over different trunk groups that terminate in servers or switches with different dial plans.
Valid Entry 0 to 28 blank
Usage Inter-Exchange Carrier (IXC) digits are deleted if the field is administered to any nonblank value. If the field is blank, no IXC digits are deleted. Communication Manager administers the IXC digits on the Inter-Exchange Carrier (IXC) Codes screen.
Use for calls that route:
· To or through a remote server.
· Over tie trunks to a private network server.
· Over local telephone company central office (CO) trunks to the serving CO.
No. Dgts Subaddress
Available only if ISDN Feature Plus is enabled for the system.
Valid Entry 1 to 5 blank
Usage
The number of dialed digits to send in the calling party subaddress IE. Allows a caller to reach a number where the media server's digit processing deletes the dialed number and inserts the listed directory number (LDN). The LDN then is sent to the destination address and the dialed extension is sent in the calling party subaddress information element (IE). At the receiving end, the call terminates to the user indicated by the subaddress number instead of the attendant.
Related links ISDN Feature Plus on page 836
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NPA
NPA
The three-digit Numbering Plan Area (NPA) or area code for the terminating endpoint of the trunk group. Not required for AAR. The local telephone company can verify this number. For WATS trunks, the terminating NPA is the same as the home NPA unless the Local Exchange Carrier requires 10 digits for local NPA calls. Leave blank for AAR calls and for tie trunks.
Numbering Format
Use this field to specify the numbering format of calls that Communication Manager routes over trunk group of the preferred routing pattern. This field applies to ISDN and SIP trunk groups.
Valid entry blank
natl-pub intl-pub locl-pub pub-unk lev0-pvt lev1-pvt lev2-pvt unk-unk
Numbering Plan Identifier Derived from the value of the Call Type field on the ARS Analysis screen or the AAR Analysis screen.
E.164(1) E.164(1) E.164(1) E.164(1) Private Numbering Plan - PNP(9) Private Numbering Plan - PNP(9) Private Numbering Plan - PNP(9) unknown(0)
Type of Numbering
Derived from the value of the Call Type field on the ARS Analysis screen or the AAR Analysis screen.
national(2)
international(1)
local/subscriber(4)
unknown(0)
local(4)
Regional Level 1(2)
Regional Level 2(1)
unknown(0)
Note:
To gain access to Telcordia Technologies NI-2 Operator Service Access, type unk-unk in the Inserted Digits field.
For Network Call Redirection or Transfer, type lev0-pvt in the Numbering Format field.
Pattern Name
An alphanumeric name for the route pattern.
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Route Pattern
Pattern Number
The route pattern number.
Prefix Mark
Sets the requirements for sending a prefix digit 1, indicating a long-distance call. Prefix Marks apply to 7- or 10-digit Direct Distance Dialing (DDD) public network calls. A prefix digit 1 is sent only when call type is foreign number plan area (FNPA) or home numbering plan area (HNPA) in the ARS Digit Analysis table.
Not required for AAR. ARS requires a number from 0 to 4 or blank.
For a WATS trunk, the Prefix Mark is the same as the local telephone company central office (CO) trunk.
Valid Entry 0
1 2 3
4 blank
Usage · Suppresses a user-dialed prefix digit 1 for 10-digit FNPA calls.
· Leaves a user-dialed prefix digit 1 for 7-digit HNPA calls.
· Leaves a prefix digit 1 on 10-digit calls that are not FNPA or HNPA calls.
Should not be used in those areas where all long-distance calls must be dialed as 1+10 digits. Check with the local network provider. Sends a 1 on 10-digit calls, but not on 7-digit calls.
Used for HNPA calls that require a 1 to indicate long-distance calls. Sends a 1 on all 10-digit and 7-digit long-distance calls.
Refers to a Toll Table to define long-distance codes. Sends a 1 on all long-distance calls and keep or insert the NPA (area code) so that all long-distance calls are 10-digit calls. The NPA is inserted when a user dials a prefix digit 1 plus 7 digits.
Refers to a Toll Table to define long-distance codes. Always suppress a user-dialed prefix digit 1.
Used, for example, when ISDN calls route to a server that rejects calls with a prefix digit 1. For tie trunks.
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SCCAN
SCCAN
If enabled, indicates that the route pattern supports incoming SCCAN calls. Available only if Enhanced EC500 is enabled for the system. Related links Enhanced EC500 on page 833
Secure SIP
If enabled, specifies using the SIP or SIPS prefix when the call is routed to a SIP trunk preference. If SIP trunks are not specified, the call is routed over whatever trunk is specified.
Service/Feature
An identifier of the Service/Feature carried by the information element (IE) in a call in this route preference. Required by Call-by-Call Service Selection, and Network Call Redirection Transfer. Accepts up to 15 characters.
Note:
User-defined service types for network facilities can also be used. Any user-defined Facility Type of 0 (feature), 1 (service), or 3 (outgoing) is allowed.
Available only if ISDN-PRI or ISDN-BRI Trunks are enabled for the system.
Valid Entry accunet i800 lds
mega800
megacom
multiquest
operator oper-meg (operator and megacon) oper-sdn (operator and sdn) outwats-bnd
sdn (allows Network Call Redirection/Transfer) sub-operator sub-op-meg
(sub-operator and megacom) sub-op-sdn
(sub-operator and sdn) wats-max-bnd
Related links Facility Type on page 630 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
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Route Pattern
Toll List
Valid Entry 1 to 32 blank
Usage
The number of the ARS Toll Table associated with the terminating NPA of the trunk group. Required for Prefix Mark 2 or 3. Not required for AAR.
Related links Prefix Mark on page 690
TSC
Use the Temporary Signaling Connection (TSC) field to support feature transparency on DCS+ calls and to support QSIG call completion. You must enable TSC to allow Communication Manager to send Call-Associated TSC (CA-TSC) and Non-Call-Associated TSC (NCA-TSC) requests for a routing preference.
Used for SIP Stations
Use the Used for SIP Stations field to set the preferred route pattern.
Valid entry Yes
No
Usage
Use this option if route pattern is to be exclusively used for SIP station routing. Communication Manager automatically populates the values of primary and secondary Session Manager node names in the Primary SM and Secondary SM fields.
The default option in No.
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Chapter 115: Security related system parameters
Determines when Avaya Communication Manager reports a security violation. Many of the fields on this screen repeat for each type of security violation. They are explained once here, but the usage is the same for all. Example command: change system-parameters security
Security related system parameters: page 1
Hide Post-Dialing DTMF On List Trace
Enable this field to display asterisk (*) characters in place of digits in the output of the list trace command. The default value of this field is n.
SECURITY VIOLATION NOTIFICATION PARAMETERS
Announcement Extension
The announcement extension where the Security Violation Notification (SVN) announcement resides. The server running Communication Manager calls the referral destination, then plays this announcement upon answer.
Originating Extension
The extension that initiates the referral call in the event of a security violation. It also sends the appropriate alerting message or display to the referral destination. If notification for more than one type of security violation is established, a different extension must be assigned to each one. When Communication Manager generates a referral call, this extension and the type of violation appear on the display at the referral destination.
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Security related system parameters
Referral Destination
The extension that receives the referral call when a security violation occurs. The referral destination telephone must have a display, unless it is an Announcement Extension. The extension can be the telephone, attendant console, or vector directory number (VDN) that receives the referral call for each type of violation. This can be the same extension for all type of violations.
The Announcement Extension field is used for a VDN. Call Vectoring Time-of-Day routing is used to route the referral call to different destinations based on the time of day or the day of the week.
Related links Announcement Extension on page 693
SVN Authorization Code Violation Notification Enabled
Enables or disables Authorization Code Violation Security Notification. By using SVN Remote Access Violation Notification Enabled, you can do the following:
· Establish parameters for remote access security violations.
· Enable or disable the feature.
A remote access violation occurs if a user enters incorrect barrier codes. The system cannot disable remote access following a security violation unless SVN Remote Access Violation Notification Enabled field is enabled.
Related links SVN Remote Access Violation Notification Enabled on page 694
SVN Login (Violation Notification, Remote Access, Authorization Code) Enabled
Enables or disables login violation notification. If enabled, Communication Manager sends a notification when a login violation occurs.
SVN Remote Access Violation Notification Enabled
Use this field to enable or disable Remote Access Violation Notification. Use with SVN Authorization Code Violation Notification Enabled to enable or disable remote access security violations. A remote access violation occurs if a user enters incorrect barrier codes. The system cannot disable remote access following a security violation unless this field is enabled.
Related links SVN Authorization Code Violation Notification Enabled on page 694
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Security related system parameters: page 2
Time Interval
Valid Entry 0:01 to 7:59
Usage
This time range, in conjunction with Login Threshold, determines if a security violation has occurred.
The range for the time interval is one minute to eight hours, entered in the screen x:xx. For example, one minute is entered as 0:01 and seven and one-half hours is entered as 7:30.
Security related system parameters: page 2
ACCESS SECURITY GATEWAY PARAMETERS
These fields are available only if Access Security Gateway (ASG) is enabled for the system. Related links
Access Security Gateway (ASG) on page 828
EPN
Indicates whether or not any entry attempt through a port that is a direct connection to the Expansion Port Network receives a challenge response.
INADS
Indicates whether or not any entry attempt through a port that is a direct connection to the Initialization and Administration System (INADS) receives a challenge response. INADS is used to remotely initialize and administer Communication Manager
MGR1
Indicates whether or not any entry attempt through a port that is a direct connection to the system administration and maintenance access interface located on the processor circuit pack receives a challenge response.
NET
Indicates whether or not any entry attempt through a port that is a dialed-in or dialed-out connection to the Network Controller circuit pack receives a challenge response.
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Security related system parameters
Translation-ID Number Mismatch Interval (days)
Valid Entry 1 to 90
Usage
The number of days the system allows access to system administration commands. When this interval expires, only init logins have the ability to execute system administration commands to modify translation data.
REMOTE MANAGED SERVICES
Port Board Security Notification
Enables or disables port board denial of service notification. Available only if RMS Feature Enabled is enabled.
Related links RMS Feature Enabled on page 696
Port Board Security Notification Interval
Valid Entry
60 to 3600 in increments of 10
Usage The interval in seconds between port board Denial of Service notifications (traps). Default is 60.
Note:
There is no delay before the first trap is sent. The interval administered in this field applies only to the period between the sending of the traps.
Available only if Remote Managed Services and Port Board Security Notification are enabled.
Related links Port Board Security Notification on page 696 REMOTE MANAGED SERVICES on page 696
RMS Feature Enabled
Enables or disables Remote Managed Services.
SECURITY VIOLATION NOTIFICATION PARAMETERS
SVN Station Security Code Violation Notification Enabled
Use this field to enable or disable the security violation notification for Station Security Codes. Station Security Codes are used to validate logins to a particular extension. The default value is n.
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Security related system parameters: page 2
STATION SECURITY CODE VERIFICATION PARAMETERS
Minimum Station Security Code Length
Valid Entry 3 to 8
Usage
The minimum required length of the station security codes. Longer codes are more secure. If station security codes are used for external access to telecommuting features, the minimum length should be seven or eight.
Receive Unencrypted from IP Endpoints
Allows or blocks unencrypted data from IP endpoints.
Security Code for Terminal Self Administration Required
Specifies whether or not a Personal Station Access code is required to enter the SelfAdministration mode. Related links Personal Station Access (PSA) on page 838
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Chapter 116: Service Hours Table
Specifies office service hours using up to 99 different tables. Available only if basic vectoring is enabled. Example command: change service-hours-table n, where n is the service hours table number.
Description
A name that provides a description of the table. Accepts up to 27 characters. The default is blank. Example Call-ahead Reservations
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Number
The table number.
Number
Start/End
The range of office hours for each day of the week.
A time is in the table from the first second of the start time (for example, 08:00:00) until the last second of the end time (for example, 17:00:59).
Valid Entry 0 to 23
0 to 59
Usage
Hour range. The hour range must be within the specified day, from 00:00 (midnight) until 23:59.
Minute range.
Use Time Adjustments from Location
The location number that specifies how time zone offset and daylight saving rule adjustments are performed.
Related links Loc Number on page 566
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Chapter 117: Signaling group
Establishes signaling group parameters for ISDN-PRI, H.323, atm and SIP trunks. Because these trunk types vary in the types of parameters needed, the system displays these fields on this screen change depending on the Group Type.
Related links Signaling group: page 1 on page 700
Signaling group: page 1
Alert Incoming SIP Crisis Calls
This field is used to enable the Emergency call routing for H.323 visiting users feature.
The Emergency call routing for H.323 visiting users feature support crisis alert for H.323 visiting users. Emergency calls made by H.323 visiting users from an enterprise network are routed to the nearest possible emergency response agencies (PSAPs) at the location visited by the user.
This field is available only for SIP signaling groups.
Valid entry y n
Usage
The attendant endpoints are notified for the incoming emergency call.
The attendant endpoints are not notified for the incoming emergency call. The default value is n.
Alternate Route Timer
Available only if Group Type is sip.
Note: You must the LAR field to a value other than none. If you set the LAR field to none, the call drops.
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Valid Entry 2 to 30
Usage
The time in seconds Communication Manager waits for a Ringing message from the far end of the trunk before trying for an alternate route to establish a call. The default value is 6 seconds.
Related links Group Type on page 708
Associated Signaling
Available only if Group Type field is isdn-pri.
Valid Entry y n
Usage Enables associated signaling. Enables non-facility associated signaling.
Related links Group Type on page 708
Bypass If IP Threshold Exceeded
Available only if Group Type is h.323 or sip.
Valid Entry y
n
Usage
Automatically remove from service the trunks assigned to this signaling group when IP transport performance falls below administered system-wide limits.
Signaling group trunks stay in service when IP transport performance falls below limits.
Related links Group Type on page 708
Calls Share IP Signaling Connection
Available only if Group Type is h.323 or sip.
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Signaling group
Valid Entry y
n
Usage
Enables inter-connection between servers running Avaya Communication Manager. Unavailable if a RAS-Location Request (LRQ) is enabled.
Note:
When the near and far-end servers are running Avaya Communication Manager, this value must match the value administered for enabling the Layer 3 test.
Disables inter-connection between servers running Avaya Communication Manager. Used if the local or remote server is not running Avaya Communication Manager.
Related links Group Type on page 708 LRQ Required on page 712 Enable Layer 3 Test on page 705
Circuit Type
Available only if Group Type is atm.
Valid Entry T1 E1
Usage U.S., Canadian, and Japanese digital transmission format. European digital transmission format.
Related links Group Type on page 708
Clustered
Valid Entry y
n
Usage Allows the signaling group to administer Cluster ID. When it is set to y, the Far-end Node Name field is replaced with Cluster ID field. Does not allow the signaling group to administer Cluster ID..
Connect
To control communications at layers 2 and 3 of the ISDN-PRI protocol, this field specifies what is on the far end of the link.
Available only if Group Type is atm.
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Valid Entry host network
pbx
Usage
The link connects Communication Manager to a computer.
The link connects Communication Manager to a local telephone company central office or any other public network switch.
The link is connected to another switch in a private network.
Related links Group Type on page 708
Country Protocol
The country protocol used by the local telephone company central office where the link terminates. This value must match the country protocol used by the far-end server. For connections to a public network, the network service provider can tell you which country protocol they are using.
Available only if Group Type is atm.
Valid Entry 1 to 25 etsi
Usage
The country code.
The network service provider uses the protocol of the European Telecommunications Standards Institute (ETSI). Used only if the Signaling Mode is isdn-pri.
Related links Signaling Mode on page 719 Country options table on page 817 Group Type on page 708
D Channel
Available only if the Group Type is atm.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
09 to 32
Usage First and second characters are the cabinet number. Third character is the carrier. Fourth and fifth character are the slot number. Six and seventh characters are the circuit number.
Related links Group Type on page 708
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Signaling group
DCP/Analog Bearer Capability
Sets the information transfer capability in a bearer capability IE of a setup message to speech or 3.1kHz. Available only if Group Type is atm or h.323.
Valid Entry 3.1kHz
speech
Usage
Provides 3.1kHz audio encoding in the information transfer capability. This is the default.
Provides speech encoding in the information transfer capability.
Related links Group Type on page 708
Direct IP-IP Audio Connections
Allows or disallows direct audio connections between H.323 endpoints. Direct audio connections save bandwidth resources and improve sound quality of voice over IP (VoIP) transmissions. For SIP trunk groups, allows direct audio connections between SIP endpoints. Available only if Group Type is h.323 or sip.
For the SIP Resiliency feature, ensure that the Direct IP-IP Audio Connections field is set to Y.
Related links Group Type on page 708
DTMF over IP
The DTMF over IP field specifies the touchtone signals that are used for dual-tone multifrequency (DTMF) telephone signaling. The DTMF over IP field is available if the Group Type field is set to h.323 or sip.
Valid Entry in-band
in-band-g711
Usage
All G711 and G729 calls pass DTMF in-band. DTMF digits encoded within existing RTP media stream for G.711/G.729 calls. G.723 is sent out-of-band. It also supports the SIP trunks.
Only G711 calls pass DTMF in-band. The system displays the in-band-g711 option if the Group Type field is set to h.323.
Table continues...
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Valid Entry out-of-band
rtp-payload
Usage
All IP calls pass DTMF out-of-band. For IP trunks, the digits are done with either Keypad IEs or H245 indications. This is the default for newly added H.323 signaling groups.
The out-of-band option enables out of band signaling for the SIP Signaling group. When you select the out-of-band option, Communication Manager sends all outgoing DTMF messages as SIP INFO messages over a SIP signaling group.
This option is interoperable with H.323 networks. You can connect an Avaya nonSIP endpoint or trunk to a voice mail system on the H.323 network that is integrated with Session Manager and Communication Manager through SIP. Examples of Avaya non-SIP endpoints or trunks are H.323, Analog, DCP, and Digital trunks.
This is the method specified by RFC 2833. This is the default value for newly added SIP signaling groups. Support for SIP trunks requires the default entry of rtppayload.
Related links Group Type on page 708
Enable Layer 3 Test
Enables or disables a Layer 3 test. Communication Manager runs the Layer 3 test to verify that all connections known at the near-end are recognized at the far-end. Available only if Group Type is H.323 and SIP. This field is enabled by default and is in read-only mode.
Note:
The Layer 3 test should be enabled for an H.323 signaling group type when the Calls Share IP Signaling Connection is enabled and the far-end is Communication Manager. Requires administration of a Far-end Node Name.
The Layer 3 Test must be enabled for the call reconstruction.
Related links Calls Share IP Signaling Connection on page 701 Far-end Node Name on page 707 Group Type on page 708
Enforce SIPS URI for SRTP
The system displays this when the Group Type field is sip. Use this field to enable or disable Communication Manager to enforce SIPS URI for any incoming SIP message with security descriptions (used for SRTP negotiation) in SDP.
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Signaling group
Valid Entry y
n
Usage
Enable Communication Manager to enforce SIPS URI for any incoming SIP message with security descriptions (used for SRTP negotiation) in SDP. Note: The system displays y as the default value for the Enforce SIPS URI for SRTP field.
Disable Communication Manager from enforcing SIPS URI for any incoming SIP message with security descriptions (used for SRTP negotiation) in SDP.
For the SIP Resiliency feature, you must set this field to n if the SIP signaling is not supported.
ETSI CCBS Support
Available only if Group Type is isdn-pri and TSC Supplementary Service Protocol is set to c for ETSI.
Valid Entry none incoming outgoing both directions
Usage Interface supports neither incoming nor outgoing ETSI CCBS. This is the default. Interface supports only incoming ETSI CCBS. Interface supports only outgoing ETSI CCBS. Interface supports incoming and outgoing ETSI CCBS.
Note:
When upgrading from a version of Communication Manager that is earlier than 5.1, this is the default.
Related links TSC Supplementary Service Protocol on page 721 Group Type on page 708
Far-end Domain
The name of the IP domain that is assigned to the far-end of the signaling group. For example, to route Session Manager calls within an enterprise, the domain assigned to the proxy server is used. For external SIP calling, the domain name could be that of the SIP service provider. Available only if Group Type is sip.
Valid Entry Character string
blank
Usage
A maximum of 40 characters used to define the name of the IP domain for which the far-end proxy is responsible, if different than the near-end domain.
Far-end domain is unspecified and the far-end IP address is used. Use if the domains are the same.
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Related links Group Type on page 708
Signaling group: page 1
Far-end Listen Port
Available only if Group Type is h.323 or sip.
Valid Entry 1 to 65535
blank
Usage
Must match the number administered for Near-end Listen Port. Typically, this is the default of 5061 for SIP over TLS.
Far-end listen port is unspecified.
Related links Near-end Listen Port on page 714 Group Type on page 708
Far-end Network Region
Available only if Group Type is h.323 or sip.
Valid Entry 1 to 2000
blank
Usage
The number of the network region that is assigned to the far-end of the signaling group. The region is used to obtain the codec set used for negotiation of trunk bearer capability.
Selects the region of the near-end node and the far-end network region is unspecified.
Related links Group Type on page 708
Far-end Node Name
Use this field to specify the node name for the far-end IP interface that is used for trunks assigned to the Signaling group. You must administer the node name on the Node Names screen.
The system displays the field only if the Group Type field is set to h.323 or sip.
Related links Node Name on page 452 Group Type on page 708
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Signaling group
Group Number
The signaling group number.
Group Type
The type of protocol used with the signaling group.
Valid Entry h.323 isdn-pri sip
Usage h.323 protocols or SBS signaling trunks Integrated Service Digital Network Primary Rate Interface For SIP
H.235 Annex H Required
Enables or disables the requirement that the Communication Manager server uses the H.235 Annex H (now called H.235.5) protocol for authentication during registration. Available only if Group Type is h.323. Requires that RAS-Location Request (LRQ) is enabled.
Related links LRQ Required on page 712 Group Type on page 708
H.245 DTMF Signal Tone Duration (msec)
Available only if DTMF over IP is set to out-of-band.
Valid Entry 80 to 350 blank
Usage
Specifies the duration of DTMF tones sent in H.245-signal messages. The default is blank.
Related links DTMF over IP on page 704 Group Type on page 708
H.323 Station Outgoing Direct Media
Available if Group Type is h.323 and Direct IP-IP Audio Connections is enabled. Also available if Group Type is SIP and Direct IP-IP Audio Connections is enabled.
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Valid Entry y
n
Usage
A call from an H.323 station over a trunk that uses this signaling group starts as a direct media call. The IP address and port of the H.323 station are sent as the media and media control channel addresses in the SETUP/INVITE message.
Note:
On an outgoing Direct Media call from an IP (H.323) telephone over this trunk, if an attempt is made to transfer the call, conference another party, or put the call on hold while the call is still in ringing state, the operation fails.
The IP address of the MEDPRO board is sent in the SETUP/INVITE message. This is the default.
Related links Direct IP-IP Audio Connections on page 704 Group Type on page 708
Idle Code
An eight-digit string that is compatible with the protocol used by the far-end switch or server. Sets the signal sent out over idle DS0 channels.
Available only if Group Type is atm.
Related links Group Type on page 708
IMS Enabled
Enables or disables accepting SIP requests that match the domain in the Far-End Domain. This field enables the signaling group to use the IMS signaling. Available only if Group Type is sip. If you are configuring Communication Manager as a feature server, set the IMS Enabled field to y. Setting this field to y only applies when Communication Manager is being configured as a feature server. If you are configuring Communication Manager as an evolution server, leave the IMS Enabled field set to n.
Related links Far-end Domain on page 706 Group Type on page 708
Incoming Dialog Loopbacks
The system displays this on the Signaling Group screen when the Group Type field is sip.
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Signaling group
Valid entries allow
eliminate
Usage
Communication Manager software connects the call and allows the SIP trunks to remain in the looparound connection. If the trunk group controlled by this SIP signaling group is used for IGAR calls, set the value allow.
Communication Manager software connects the call and eliminates the SIP trunks from the looparound connection. The default value is eliminated.
Initial IP-IP Direct Media
The system displays the Initial IP-IP Direct Media field on the Signaling Group screen only when you set the Group Type field to sip and the Direct IP-IP Audio Connections field to y. The Initial IP-IP Direct Media field enables or disables the SIP direct media feature between SIP endpoints. The Initial IP-IP Direct Media field also enables or disables the direct media feature from SIP endpoints to Avaya H.323 stations. The default value for this field is n.
Valid Entry n y
Usage Disables the SIP direct media feature. The default value is n. Enables the SIP direct media feature and SIP-H.323 direct media feature.
Note:
If Initial IP- IP Direct Media? is set to y in the SIP Signaling Group then Initial INVITE with SDP for secure calls? must also be set to y in the system parameter features forms.
For the SIP Resiliency feature, ensure that the Initial IP- IP Direct Media? and the Initial INVITE with SDP for secure calls? field must be set to y.
Related links Direct IP-IP Audio Connections on page 704 Group Type on page 708
Interface
Controls how your server negotiates glare with the far-end switch. Available only for a far-end PBX connection.
Valid entry
Usage
U.S. private network applications
network
The server overrides the other end when glare occurs. For servers connected to a host computer.
user
The server releases the contested circuit and looks for another when glare occurs.
For servers connected to a public network.
Table continues...
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Signaling group: page 1
Valid entry
Usage
Private networks, including QSIG networks, applications outside the U.S.
peer-master
The switch overrides the other end when glare occurs.
peer-slave
The switch releases the contested circuit and looks for another when glare occurs.
Related links Connect on page 702
Interface Companding
The companding algorithm expected by the system at the far end.
Valid Entry a-law mu-law
Usage Algorithm expected at the far-end for E1 service. Algorithm expected at the far-end for T1 service.
Interworking Message
Determines what message Communication Manager sends when an incoming ISDN trunk call is routed over a non-ISDN trunk group.
Valid Entry PROGress
ALERTing
Usage
Requests the public network to cut through the B-channel and let the caller hear tones such as ringback or busy tone provided over the non-ISDN trunk. Normallyselected value.
Causes the public network in many countries to play ringback tone to the caller. This value is used only if the DS1 is connected to the public network, and it is determined that callers hear silence (rather than ringback or busy tone) when a call incoming over the DS1 interworks to a non-ISDN trunk.
IP Audio Hairpinning
Enables or disables hairpinning for H.323 or SIP trunk groups. H.323 endpoints are connected through the IP circuit pack without going through the time division multiplexing (TDM) bus. Available only if Group Type is h.323 or sip.
Related links Group Type on page 708
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Signaling group
IP Video
Enables or disables IP video capability for this signaling group. Available only if the signaling group type h.323 and sip.
Link Loss Delay Timer (sec)
Specifies how long to hold the call state information in the event of an IP network failure or disruption. Communication Manager preserves calls and starts this timer at the onset of network disruption (signaling socket failure). If the signaling channel recovers before the timer expires, all call state information is preserved and the signaling channel is recovered. If the signaling channel does not recover before the timer expires, the system:
· Raises an alarm against the signaling channel
· Maintains all connections with the signaling channel
· Discards all call state information about the signaling channel
Valid Entry 1 to 180
Usage
The number of seconds to delay the reaction of the call controller to a link bounce. Default is 90.
Location for Routing Incoming Calls
The system displays this only when the Group Type field on the Signaling Group screen is h.323 and Multiple Locations field on the Optional Feature screen is y.
Valid Entry 1 to 2000
blank
Usage
(Depending on your server configuration, see Avaya Aura® Communication Manager System Capacities Table.) Determines the location-based digit analysis which is applied to the destination digits dialed by the incoming trunk call.
The location of CLAN's cabinet or the location of the Processor Ethernet (PE) is taken for routing the call, depending on whether CLAN or PE is being used to carry that H.323 signaling channel. By default, the value is blank.
LRQ Required
Enables or disables RAS-Location Request (LRQ) messages. Available only if Group Type is h.323.
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Signaling group: page 1
Valid Entry y
n
Usage
Allows IP trunk availability to be determined on a per call basis. An LRQ message is sent to the far-end gatekeeper prior to each call over the IP trunk. The far-end gatekeeper responds with a RAS-Location Confirm (LCF) message, and the call proceeds. Required if H.235 Annex H Required is enabled. Calls Share IP Signaling Connection must be disabled.
For far-end servers running Communication Manager, unless H.235 Annex H Required is enabled.
Related links H.235 Annex H Required on page 708 Calls Share IP Signaling Connection on page 701 Group Type on page 708
Max number of CA TSC
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry 0 to 619
Usage
The maximum number of simultaneous call-associated Temporary Signaling Connections that can exist in the signaling group. Typically this is the number of ISDN-PRI trunk group members controlled by this signaling group.
Related links Group Type on page 708
Max number of NCA TSC
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry 0 to 256
Usage
The maximum number of simultaneous non-call-associated Temporary Signaling Connections. The TSCs carry signaling for features not associated with a specific call, for example, signals to turn on Leave Word Calling.
Related links Group Type on page 708
Media Encryption
Enables or disables encryption for trunk calls assigned to this signaling group. If encryption for the signaling group is disabled, then trunk calls using this signaling group do not get encrypted
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Signaling group
regardless of IP Codec Set administration. Available only if Media Encryption is enabled in Communication Manager and the Group Type is h.323. Related links Group Type on page 708
Name
A name that identifies the signaling group. Accepts up to 15 alphanumeric characters.
Note: Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly. Related links Group Type on page 708
Near-end Listen Port
Available only if Group Type is h.323 or sip.
Valid Entry 5000 to 9999 1719 1720
Usage An unused port number for the near-end listen port. For LRQ-enabled ports For h.323-enabled ports
Related links H.323 Station Outgoing Direct Media on page 708 LRQ Required on page 712 Group Type on page 708
Near-end Node Name
The node name for the Control LAN (C-LAN) IP interface in the Avaya S8XXX Server. The node name must be previously administered. Available only if Group Type is h.323 or sip.
Note: This field can be set to procr or procr6, for IPv4 and IPv6 signaling. Related links Name on page 475
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Node Name on page 452 Group Type on page 708
Signaling group: page 1
Network Call Transfer
Enables or disables Network Call Transfer so that D-channels support Explicit Network Call Transfer (ENCT). Available only if Group Type is atm.
Related links Group Type on page 708
Passphrase
The passphrase used to generate a shared "secret" for symmetric encryption of the media session key. The same passphrase must be assigned to the corresponding signaling groups at both ends of an IP trunk. The passphrase:
· Consists of 8 to 30 alphanumeric characters
· Is case sensitive
· Must contain at least one alphabetic and at least one numeric
· Valid characters also include letters, numbers, and these symbols: !&*?;'^(),.:-
The passphrase is used for both Media Encryption and authentication. This field cannot be left blank.
Available only if Media Encryption is enabled or the H.235 Annex H Required is enabled.
Related links H.235 Annex H Required on page 708
Peer Detection Enabled
The values for this field are n(o) and y(es). The default value is y(es). This field is only available if Group Type is sip. When set to y(es) then Communication Manager will automatically set the value in the Peer Server field when the sip signaling group comes into service.
Peer Server
The values for this field are Others, SM, and AMS. The default value is Others. This field is only available if Group Type is sip. If this sip signaling group terminates to an Avaya Session
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Signaling group
Manager, then Communication Manager will automatically set value in this field to SM if the Peer Detection Enabled field is set to y(es). If Clustered field is set to y, then the value in this field is automatically changed to SM, and this field will be in read-only mode.
Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers
If you set the Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers field on the SIP Signaling Group screen to y and if Communication Manager uses the publicunknown-numbering table, Communication Manager inserts a leading plus (+) digit in calling, alerting, diverting, or connected numbers. The default value is n.
If the Peer Server field on the SIP Signaling Group screen is SM, the system sets the Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers field to display-only, with the default value of y. If the Peer Server field on the SIP Signaling Group screen is Others, the system sets the Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers field to n and as editable.
Primary D Channel
Available only if Group Type is isdn-pri.
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
09 to 32
Usage First and second characters are the cabinet number. Third character is the carrier. Fourth and fifth character are the slot number. Six and seventh characters are the circuit number.
Related links Group Type on page 708
Priority Video
Enables or disables the priority video function that specifies that incoming video calls have an increased likelihood of receiving bandwidth, and are also allocated a larger maximum bandwidth per call.
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Available only if:
· Group Type is h.323 or sip
· Multimedia SIP trunking is enabled for the system
Related links Multimedia IP SIP Trunking on page 837 Group Type on page 708
Signaling group: page 1
Protocol Version
Available only if Group Type is atm.
Valid Entry a b c d
Usage
In countries whose public networks allow multiple layer-3 signaling protocols for ISDN-PRI service. Selects the protocol that matches the network service provider's protocol.
Related links Group Type on page 708
Q-SIP
The system displays this only when the Group Type field is h.323 or sip.
Valid Entry y n
Usage Enables the QSIG over SIP (Q-SIP) feature for the signaling group. The QSIG over SIP feature is disabled. By default, the value is n.
QSIG Signaling Group
The system displays this only when the Group Type field is sip and the Q-SIP field is set to y.
Valid Entry 1 to 999 blank
Usage Assigns a number for the QSIG signaling group. No QSIG signaling group is assigned. By default, the value is blank.
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Signaling group
Remove '+' from Incoming Called/Calling/Alerting/Diverting/ Connected Numbers
If you set the Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field on the SIP Signaling Group screen to y, Communication Manager removes the leading plus (+) digit, if present, from both calling and called numbers. The default value is y.
This field interacts with the Peer Server field in the following manner:
· If the Peer Server field on the SIP Signaling Group screen is SM, the system sets the Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field to n. The Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field is editable.
· If the Peer Server field on the SIP Signaling Group screen is Others, the system sets the Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field to y. The Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers field is editable.
Remote Office
Enables or disables administering this signaling group for a remote office. Available only if Group Type is h.323.
Related links Group Type on page 708
RFC 3389 Comfort Noise
The system displays this on the Signaling Group screen when the Group Type field is sip.
Valid entries y
n
Usage
This enables SIP signaling for comfort noise. If RFC 3389 Comfort Noise field is set to y, this field overrides the Silence Suppression field on the IP Codec Set screen.
This disables SIP signaling for comfort noise. Default is n.
RRQ Required
Enables or disables the requirement that a vendor registration be sent. Available only if Group Type is h.323.
Related links Group Type on page 708
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Signaling group: page 1
SBS
Enables or disables the Separation of Bearer and Signaling (SBS) trunk groups. If this field is enabled, both the Trunk Group for NCA TSC and the Trunk Group for Channel Selection must be set to the signaling group number administered for the SBS trunk group. Available only if Group Type is set to h.323.
Related links Trunk Group for NCA TSC on page 720 Trunk Group for Channel Selection on page 720 Group Type on page 708
Session Establishment Timer (min)
Use this field to specify the time in minutes that Communication Manager waits before tearing down a ring no answer call. You can enter a value from 3 to 240. The default value is 3.
The system displays this field only when the Group Type field is set to sip.
Related links Group Type on page 708
Signaling Mode
Displays the isdn-pri signaling mode. Available only if Group Type is atm. Related links
Group Type on page 708
SIP Signaling Group
The system displays this only when the Group Type field is h.323 and the Q-SIP field is set to y.
Valid Entry 1 to 999 blank
Usage Assigns a number for the SIP signaling group. No SIP signaling group is assigned. By default, the value is blank.
T303 Timer (sec)
Available only if Group Type is h.323.
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Signaling group
Valid Entry 2 to 10
Usage
The number of seconds the system waits for a response from the far end before invoking Look Ahead Routing. Default is 10.
Related links Group Type on page 708
Transport Method
Available only if Group Type is sip.
Valid Entry tcp tls
Usage Transport is accomplished using Transmission Control Protocol (TCP). Transport is accomplished using Transport Layer Security (TLS). This is the default.
For the SIP Resiliency feature, ensure that the TLS is selected as a transport method.
Related links Group Type on page 708
Trunk Group for Channel Selection
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry 1 to 2000
Usage Trunk group number used for channel selection.
Related links Group Type on page 708
Trunk Group for NCA TSC
Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry 1 to 2000 blank
Usage Trunk group number used for non-call-associated Temporary Signaling Connections.
Related links Group Type on page 708
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Signaling group: page 1
TSC Supplementary Service Protocol
The supplementary service protocol used for temporary signaling connections. Available only if Group Type is atm, h.323, or isdn-pri.
Valid Entry a b c d e
f g
Usage AT&T, Telcordia Technologies, Nortel. ISO QSIG. Also, for SBS signaling groups. ETSI. Available only if Group Type is isdn-pri. ECMA QSIG Allows DCS with rerouting. DCS with Rerouting must be enabled, and Used for DCS must be enabled for the trunk group. Feature Plus ANSI. Available only if ISDN-PRI, or ISDN-BRI, or Used for DCS are enabled for the system.
Related links DCS with Rerouting on page 832 Used for DCS on page 953 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836 Group Type on page 708
Virtual Channel Identifier
Available only if Group Type is atm.
Valid Entry 32 to 1023 blank
Usage The number used as a Virtual Channel Identifier.
Related links Group Type on page 708
Virtual Path Identifier
Available only if Group Type is atm.
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Signaling group
Valid Entry 32 to 1023 blank
Usage The number used as a Virtual Path Identifier.
Related links Group Type on page 708
X-Mobility/Wireless Type
The type of X-Mobile endpoints allowed. Available only if Group Type is isdn-pri.
Valid Entry DECT
none
Usage
The remote end of the trunk group controlled by the signaling group is a DECT mobility controller. This allows X-Mobility to work over ISDN-PRI trunks between the server/switch and adjunct.
Not administered.
Related links Group Type on page 708
Signaling group: page 2
This screen must be filled in for ATM signaling groups. It provides two functions: · Defines fractional T1 and fractional E1 facilities, specifying how many and which channels to use. · Specifies the port numbers to use. Port numbers must be unique for all signaling boards on the same ATM board.
Available only if Group Type is atm.
LIMIT SIGNALING GROUP USAGE
The LIMIT SIGNALING GROUP USAGE screen is available only if the Group Type field is set to h.323 or sip, and the Near-end Node Name field is set to procr. Use this screen to control where H.323 and SIP trunks are used. Related links Group Type on page 708 Near-end Node Name on page 714
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Signaling group: page 2
Enable on Survivable Processors (ESS and LSP)
Valid Entry all
ess-all
none selected
Usage
To enable the signaling group and the associated trunks on both - Survivable core server (Enterprise Survivable Server (ESS)) and Survivable remote server (Local Survivable Processor (LSP)).
To enable the signaling group and the associated trunks on Survivable core server (Enterprise Survivable Server) but not on Survivable remote server (Local Survivable Processor).
To block the signaling group and the associated trunks on survivable processors and free system resources.
To specify which survivable processors can use the signaling group and the associated trunks.
Enable on the Main Processor(s)
Use the Enable on the Main Processor(s) field to enable or disable the signaling group and the associated trunks on the main server.
Valid Entry y n
Usage To enable the signaling group and the associated trunks on the main server. To disable the signaling group and the associated trunks on the main server.
Selected Survivable Processor Node Names
The Selected Survivable Processor Node Names field is available only if you set the Enable on Survivable Processors (ESS and LSP) field to selected. Use this field to specify the node names of the survivable processors that can use the signaling group and the associated trunks.
Related links Enable on Survivable Processors (ESS and LSP) on page 723
SIGNALING GROUP
Chan Port
Valid Entry 009 to 256 blank
Usage
The port number for non-signaling channels.
The signaling channel (port 16 for an E1 and port 24 for a T1) must be a port between 9 and 32. A port number used here cannot be used on any other ATM signaling group on the same board.
The channels used must match exactly the channels used on the other end of the signaling group. For example, if your T1 is set up to use channels 1 through 5, 7, and 24 (the signaling channel), the far end must also use channels 1 through 5, 7, and 24.
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Signaling group
Enable Failover Event Package Subscription
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
This field is available only when you set the Multiple Level Precedence & Preemption field on the System parameters customer options screen to y.
For Communication Manager to send and receive failover notify messages, you must configure CCA-ID on the Multiple Level Precedence & Preemption screen.
Note:
MLPP is a customer option. Therefore, you can change the parameters only through a license change.
Valid Entry
y n
Usage
over Displays the option request parameters on the screen. Does not display the option request parameters on the screen. The default value is n.
Failover/failback Signaling-group Pair
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
Type the Signaling group that connects to the primary Multi-Function Soft Switch (MFSS) or secondary MFSS.
A lower signaling group number is always the number assigned to the primary MFSS from the pair of signaling groups.
Valid Entry SIP signaling-group number Blank
Usage The number that connects to the far-end MFSS. The field is blank by default.
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Signaling group: page 2
OPTIONS Request Parameters
Note: This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers. The system displays the OPTIONS Request Parameters fields on the screen only when you set the Enable Failover Package Subscription field to y. Communication Manager creates and sends periodic OPTIONS messages to the MFSS. OPTIONS Request Parameters on the Signaling group screen monitor the interval between OPTIONS messages.
Interval between OPTIONS messages (seconds)
The interval time for Communication Manager to send OPTIONS requests to MFSS. You can enter a number from 30 to 900 in increments of 10. The default value is 60.
Number of OPTIONS message failure before failover
The number of OPTIONS messages that Communication Manager sends to the far-end MFSS without getting a response. A failover is triggered when the far-end MFSS does not respond to the OPTIONS messages. You can enter a number from 1 to 10. The default value is 2.
Number of OPTIONS message successes before failback
The number of successful OPTIONS messages that Communication Manager sends to the farend MFSS. A failback is triggered when the far-end MFSS starts responding to the OPTIONS messages. You can enter a number from 1 to 10. The default value is 2.
Wait time before failback (seconds)
The time taken for Communication Manager to failback to the primary MFSS after the far-end MFSS starts responding to the OPTIONS messages. You can enter 1 or any number from 10 to 250 in increments of 10.
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Signaling group
Signaling group: NCA-TSC Assignment page
Appl.
Specifies the application for this administered NCA-TSC.
Valid Entry audix dcs gateway masi qsig-mwi
Usage The ISDN-PRI D-channel DCS Audix feature The DCS Over ISDN-PRI D-channel feature. The administered NCA-TSC is used as one end in the gateway channel. The NCA-TSC is one end of a multimedia application server interface. Message conversion from an administered AUDIX NCA-TSC to a QSIG CISC. A Machine ID between 1 and 20 is required.
As-needed Inactivity Time-out (min)
Valid Entry 10 to 90 blank
Usage Applies only to as-needed NCA-TSCs.
Dest. Digits
An ISDN interface extension number. Accepts up to 15 characters including * and #.
Enabled
Enables or disables the administered NCA-TSC.
Established
The strategy for establishing this administered NCA-TSC.
Valid Entry permanent
as-needed
Usage
The administered NCA-TSC can be established by either the near end or the far end.
The administered NCA-TSC is established the first time the administered NCA-TSC is needed. The NCA-TSC can be set up either by the near end or far end switch.
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Signaling group: NCA-TSC Assignment page
Local Ext
The local extension of the ISDN interface.
Mach ID
Valid Entry
1 to 63 for DCS, 1 to 99 for AUDIX, 1 to 15 for MASI, and 1 to 63 for QSIGMWI
Usage
A unique machine ID. You cannot specify an ID that you already administered for the Processor Channel.
Related links Mach ID on page 668
Service/Feature
Valid Entry accunet i800 inwats lds mega800 megacom multiquest operator sdn sub-operator wats-max-band
Usage
The assigned service or feature. In addition to pre-defined services or features, any user-defined Facility Type of 0 (feature) or 1 (service) is allowed.
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Chapter 118: SIT Treatment for Call Classification
The treatment of Special Information Tones (SITs) used for Outbound Call Management type calls with USA tone characteristics. The port network TN744 Call Classifier circuit pack ports or Branch Gateway internal tone detector resources in classified mode are used to detect SITs. The classifiers are capable of detecting the following SITs:
· SIT Ineffective Other · SIT Intercept · SIT No Circuit · SIT Reorder · SIT Vacant Code · SIT Unknown · AMD (Answering Machine Detected) Treatment Available only if ASAI Link Core Capabilities and ASAI Link Plus Capabilities are enabled for the system. Example command: change sit-treatment
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Related links ASAI Link Core Capabilities on page 829 ASAI Link Plus Capabilities on page 829
AMD Treatment
AMD Treatment
Specifies the treatment for Answering Machine Detected. An ASAI adjunct can request AMD for a call. If an Answering Machine is detected, one of two treatments is specified.
Valid Entry answered dropped
Usage
Calls are classified as answered, and are therefore sent to an agent.
Calls are classified as unanswered, and are therefore not sent to an agent. This is the default.
Pause Duration
Fractions of a second pause duration, as opposed to Talk Duration that is for full seconds.
Valid Entry
Usage
0.1 to 2.0 seconds The amount of time Communication Manager looks for a pause before it classifies
in increments of 0.1 the call as a live person.
seconds
The Pause Duration timer should be set longer than the typical silence between
words in an answering machine greeting, but shorter than the typical space between
words in a live greeting.
Defaults to 0.5 seconds.
SIT Ineffective Other
The treatment for "Ineffective Other" messages. An example of an Ineffective Other message is "You are not required to dial a 1 when calling this number".
Valid Entry answered dropped
Usage
Calls are classified as answered, and are therefore sent to an agent.
Calls are classified as unanswered, and are therefore not sent to an agent. This is the default.
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SIT Treatment for Call Classification
SIT Intercept
The treatment for intercept messages that direct callers to another number or extension.
Valid Entry answered
dropped
Usage
Calls are classified as answered, and are therefore sent to an agent. This is the default.
Calls are classified as unanswered, and are therefore not sent to an agent.
Example An example of an intercept message is "XXX-XXXX has been changed to YYY-YYYY, make a note of it".
SIT No Circuit
The treatment for calls with circuit problems, such as busy.
Valid Entry answered dropped
Usage
Calls are classified as answered, and are therefore sent to an agent.
Calls are classified as unanswered, and are therefore not sent to an agent. This is the default.
SIT Reorder
The treatment for reorder calls.
Valid Entry answered dropped
Usage
Calls are classified as answered, and are therefore sent to an agent.
Calls are classified as unanswered, and are therefore not sent to an agent. This is the default.
SIT Unknown
The treatment for calls when a situation or condition that is unknown to the network is encountered.
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Valid Entry answered dropped
SIT Vacant Code
Usage Calls are classified as answered, and are therefore sent to an agent. Calls are classified as unanswered, and are therefore not sent to an agent. This is the default.
SIT Vacant Code
The treatment for calls when a vacant code is encountered.
Valid Entry answered dropped
Usage
Calls are classified as answered, and are therefore sent to an agent.
Calls are classified as unanswered, and are therefore not sent to an agent. This is the default.
Talk Duration
Full second talk duration, as opposed to Pause Duration that is for fractions of a second.
Valid Entry
Usage
0.1 to 5.0 seconds The amount of time Communication Manager looks for voice energy. If it finds that
in increments of 0.1 much continuous speech, Communication Manager classifies the call as an
seconds
answering machine.
The Talk Duration timer should be set to a time longer than it takes to say a typical live greeting.
Defaults to 2.0 seconds.
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Chapter 119: Site Data
Provides descriptive information about the buildings, floors, and telephone set colors. Values must be supplied here before the site data can be entered for the station. From Release 8.1.3.1 onwards, the Site Data screen displays 14 pages of building strings that can hold 1000 entries, instead of two pages containing 150 entries. To modify the fields on the Site Data screen, run the following command: change site-data.
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Chapter 120: Station
Administers individual telephone sets or virtual telephones. Some of the fields are used for specific telephone types, while others are used for all telephone types. To change the value of the fields on the Station screen, run the following command: change station n, where n is the station number. To add a station, run the following command: add station next
The following table lists various set types. Note: The table also contains older set types. However, Avaya does not provide support for End of Manufacturers Support (EoMS) set types. For more information on EoMS set types, see https:// downloads.avaya.com/css/P8/documents/100172510.
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Station
Telephone type Single-line analog
CallerID
Single-line DS1/DSO (Lineside T1/DS1) Terminals Multi-appearance hybrid
Model 500 2500 and 2500 with Message Waiting Adjunct 6210 6211 6218 6219 6220 6221 Analog telephone with caller ID 7101A and 7102A 7103A Programmable and Original 7104A 8110 DS1FD 7302H, 7303H Voice Response Unit (VRU) with C and D tones VRU without C and D tones DS1 device without forward disconnect VRU with forward disconnect without C and D tones VRU with forward disconnect without C and D tones 510D 515BCT 7303S 7305H 7305S 7309H 7313H 7314H 7315H 7316H 7317H
Administered as 500 2500
6210 6210 6218 6218 6220 6220 CallrID 7101A 7103A 7104A 8110 DS1FD 7303S VRU
2500 OPS DS1FD or DS1SA
VRUFD or VRUSA
510 515 7303S, 7313H 7305S 7305S, 7316H, 7317H 7309H, 7313H 7313H 7314H 7315H 7316H 7317H
Table continues...
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Telephone type
Multi-appearance digital
Model 2402 2410 2420 6402 6402D 6408 6408+ 6408D 6408D+ 6416D+ 6424D+ 7401D 7401+ 7403D 7404D 7405D 7406D 7406+ 7407D 7407+ 7410D 7410+ 7434D 7444D 8403B 8405B 8405B+ 8405D 8405D+ 8410B 8410D 8411B 8411D 9408 9404
Administered as 2402 2410 2420 6402 6402D 6408 6408+ 6408D 6408D+ 6416D+ 6424D+ 7401D 7401+ 7403D 7404D 7405D 7406D 7406+ 7407D 7407+ 7410D 7410+ 7434D 7444D 8403B 8405B 8405B+ 8405D 8405D+ 8410B 8410D 8411B 8411D 9408 9404
Table continues...
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Station
Telephone type IP Telephone
Model CALLMASTER I CALLMASTER II, CALLMASTER III, CALLMASTER IV CALLMASTER VI IDT1 IDT2 4601+
Note:
When adding a new 4601 IP telephone, you must use the 4601+ station type. This station type has the Automatic Callback feature enabled. 4602+
Note:
When adding a new 4602 IP telephone, you must use the 4602+ station type. This station type has the Automatic Callback feature enabled. 4606 4610 4612 4620SW IP with G3.5 hardware 4621 4622 4624 4625 4690 9611 9621 9630 9641 9650
Administered as 602A1 603A1, 603D1, 603E1, 603F1
606A1 7403D 7406D 4601+
4602+
4606 4610 4612 4620 4621 4622 4624 4625 4690 9611 9621 9630 9641 9650
Table continues...
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Telephone type SIP IP Telephone
Model · 4602SIP with SIP firmware · 4610SIP with SIP firmware · 9608SIPCC · 9611SIPCC · 9621SIPCC · 9641SIPCC · 4620SIP with SIP firmware · SIP Softphone or Avaya one-X Desktop · Toshiba SP-1020A
Administered as 4620SIP
Note:
Any model telephone with SIP firmware used for SIP networking must be administered as a 4620SIP, 96xxSIP, or 16CC telephone.
Note:
Communication Manager Release 6.2 and later do not support 1616SIP CC and 4620SIP CC telephones.
9601SIP
9620, 9630, 9630G, 9640, 9640G with SIP firmware
9608SIP 96xx or 96xxSIP telephone
Note:
· Communication Manager Release 8.0.1 and later internally maps the Avaya Conference Phone B199 conference phones with 9630SIP set types.
· You can administer Avaya Conference Phone B199 conference phones from the System Manager user interface only.
9608 with SIP firmware
9611 with SIP firmware
9621 with SIP firmware
9641 with SIP firmware
Avaya J129 IP Phone
-- 9611SIP 9621SIP 9641SIP J129
Table continues...
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Station
Telephone type
H.323 SoftPhone
Personal computer Voice or data Test line Administered without hardware Key telephone system interface Administered without hardware
CTI XMOBILE ISDN-BRI data module Separation of Bearer and Signaling (SBS) extension
Model Avaya J169 IP Phone Avaya J179 IP Phone J169CC IP Phone J179CC IP Phone Road Warrior application Native H.323 Single connect Any NI-BRI N1 and N2 telephone 7505D 7506D 7507D 8503D 8510T 8520T 6300 or 7300 6538 or 6539 ATMS --
--
any digital set Computer telephony integration (CTI) station CTI station EC500, DECT, or PHS 7500
SBS test extension without hardware
Administered as J169 J179 J169CC J179CC H.323 or DCP type H.323 H.323 or DCP type NI-BRI 7505D 7506D 7507D 8503D 8510T 8520T Personal computer Constellation 105TL XDID or XDIDVIP
K2500
-- CTI
CTI XMOBILE 7500
SBS
Table continues...
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Telephone type CS1K Endpoints
Model · CS1k-39xx · CS1k-1col · CS1k-2col · CS1k-ana · CS1k-IP · CS1k-IPCC
Administered as CS1k
Station: Page 1
Station: Page 1
Attendant
Use this field to configure SIP attendant. This field appears for set type 9641SIP only.
Valid Entry n y
Usage
Disables SIP attendant. This is the default value.
Enables SIP attendant.
When you set the SIP value to y, ensure that:
· Other stations should not have Bridge Appearance of this station.
· This station should not be part of any group. For example pickup group, hunt group, or termextension group.
· This station should not have coverage path configured.
· Call diversion feature should not be enabled on stations. For example, SAC and call forward
· Call restriction feature should not be enabled on stations. For example LNCC and Restrict Last Appearance
BCC
Indicates voice or voice-grade data when the value is set to 0. The Bearer Capability Classes (BCC) value is used to determine compatibility when non-ISDN facilities are connected to ISDN facilities (ISDN Interworking). Available only if ISDN-PRI or ISDN-BRI Trunks are enabled for the system.
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Station
Bearer
Used when Secure Terminal Equipment (STE) telephones are administered as 8510 telephones. The system the Bearer field for 8503, 8510, and 8520 stations in Communication Manager 2.1 and 2.2 only. Secure Terminal Equip is for Bearer functionality in Communication Manager 3.0 and later.
Valid Entry speech
3.1khz
Usage
Forces the Bearer Cap IE to "speech" before a call is delivered to the 85xx BRI station.
Leaves the Bearer Cap IE unchanged. Use 3.1khz to let secure calls from Secure Terminal Equipment (STE) telephones to work properly.
Button Module Pages
This field can be configured for CS1000 station types only.
Buttons per Page
This field can be configured for CS1000 station types only. This field appears if Button Module Pages field is set to a non-zero value. This field displays a default value of 24 and allows the values 8, 12, 18, 22, and 24. If the value in this field is lower than 24, Communication Manager will do one of the following:
· Allows the Button Module Page to display 24 buttons. However, it blocks form submission if there is a non-blank button in a position higher than the configured number.
· Reduce the number of button fields on the Button Module Pages to the configured number.
Coverage Path 1 or Coverage Path 2
The coverage-path number or time-of-day table number assigned to the station.
Note: If Modified Misoperation is active, a Coverage Path must be assigned to all stations on Communication Manager.
Cell Phone Number
This field can be configured when the set type is XMOBILE. The unformatted cell telephones published external numbers consisting of 1 to 15 digits. This field can contain a three-digit area
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Station: Page 1
code plus the seven-digit main number. If the same Cell Phone Number is administered for multiple XMOBILE stations, then the Dial Prefix associated with each instance of the Cell Phone Number must be the same. This number consist of a full 10-digit Cell Phone Number regardless of whether the cell telephone is local or not.
Caller ID Message Waiting Indication
Allows or prevents aliasing of various non-Avaya telephones and adjuncts. Available only if Type is CallrID. For CallrID type telephones or analog telephones with Caller ID adjuncts only.
Note: The Caller ID Message Waiting Indication administration is independent of the administration of LED or NEON-lamp Communication Manager Message Waiting Indication (MWI). For example, it is possible to administer a Caller ID telephone with Caller ID Message Waiting Indication is disabled and Message Waiting Indicator set to neon.
Customizable Labels
Enables or disables the Increase Text for Feature Buttons feature for this station. This feature expands the text labels associated with Abbreviated Dial buttons from the current five uppercase alphanumeric characters to a maximum of 13 upper and lower case alphanumeric characters. Ensures that there will always be sufficient customized button resources to support VIP users. Available only when the station type is one of the following:
· 2410 (Release 2 or later) · 2420 (Release 4 or later) · 4610 (IP Telephone Release 2.2 or later) · 4620 (IP Telephone Release 2.2 or later) · 4621 (IP Telephone Release 2.2 or later) · 4622 (IP Telephone Release 2.2 or later) · 4625 (IP Telephone Release 3.1 or later)
COR
Class of Restriction (COR) number with the required restriction.
COS
The Class of Service (COS) number used to select allowed features.
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Station
Configuration Set
Configuration Set field can be configured when the set type is XMOBILE.
Valid Entry 1 to 10
blank
Usage
The configuration set number that contains the call treatment options required for the XMOBILE station. This field must be administered if:
· The XMOBILE Type is EC500.
· The Mobility Trunk Group is a trunk group number and the administered trunk group is non-DECT or non-PHS.
· The Mobility Trunk Group is aar or ars.
If the Mobility Trunk Group is a trunk group number and the administered trunk group is DECT or PHS, this field can be left blank.
Calls Allowed
Identifies the Extension to Cellular call filter type for an XMOBILE station. This field allows an XMOBILE station to function as a bridge and still be restricted. Available only with an EC500-type XMOBILE station and with a Mapping Mode of termination or both.
Valid Entry internal
external all none
Usage
External calls are blocked. Internal calls terminate to the XMOBILE station. Attendant-originated and attendant-delivered calls are not delivered
Internal calls are blocked. External calls terminate to the XMOBILE station.
All calls terminate to the XMOBILE station.
Prevents calls from terminating to the XMOBILE station. Can be used to prevent business-related calls from accruing telephone charges on cellular telephones that are lost, being transferred to a new user, or being disabled for other business reasons.
Note: Interswitch calls on DCS trunks are treated as internal calls. When this field is set to:
· internal or all, DCS calls are delivered to the cell telephone · external or none, DCS calls are not delivered Incoming calls from other Extension to Cellular users are internal if office caller ID is enabled for the XMOBILE station associated with the cell telephone. When this field is set to: · internal or all, calls from other Extension to Cellular users are delivered · external or none, calls from other Extension to Cellular users are not delivered
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Station: Page 1
Dial Prefix
This field can be configured when the set type is XMOBILE. The unformatted sequence of up to four digits or characters that are prepended to the cell telephone's published cell telephone number before dialing. Accepts the * and # characters. If the same Cell Phone Number is administered on multiple XMOBILE stations, then the Dial Prefix associated with each instance of the Cell Phone Number must be the same.
Data Module
Indicates whether or not this telephone has an associated data module.
Display Caller ID
Enables or disables transmission of calling party information to the Caller ID telephone or adjunct. For CallrID type telephones or analog telephones with Caller ID adjuncts only. Available only if the station type is CallrID.
Display Cartridge
Enables or disables displaying the cartridge associated with the station. For 7404 D telephones only.
Display Language
Valid Entry english french italian spanish user-defined
Usage The language that displays on stations. Time of day is displayed in 24-hour format (00:00 - 23:59) for all languages except English, which is displayed in 12-hour format (12:00 a.m. to 11:59 p.m.).
Table continues...
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Station
Valid Entry unicode
Usage
Displays English messages in a 24-hour format. If no Unicode file is installed, displays messages in English by default.
Note:
Unicode display is only available for Unicode-supported telephones. Currently, 4610SW, 4620SW, 4621SW, 4622SW, 16xx, 96xx, 96x1, 9600-series telephones (Avaya one-X Deskphone Edition SIP R2 or later), and Avaya J100 Series IP Phones support Unicode display. Unicode is also an option for DP1020 (aka 2420J) and SP1020 (Toshiba SIP Phone) telephones when enabled for the system.
Extension
The extension for this station.
For a virtual extension, a valid physical extension or a blank can be entered. With blank, an incoming call to the virtual extension can be redirected to the virtual extension "busy" or "all" coverage path.
The extension length must be within 16 digits.
Endpt Init
Indicates the terminal's endpoint initialization capability. Endpoint initialization is a procedure, required for multipoint operation, by which User Service Order Profile (USOP) is associated with an endpoint on the ISDN-BRI. This association is made through the SPID, administered into the system, and entered into the ISDN-BRI terminal.
Available only if MIM Support is enabled.
Valid Entry y
n
Usage
The terminal supports Telcordia Technologies ISDN-1 terminal initialization procedures.
For all other country protocols.
Expansion Module
Indicates whether or not this telephone has an expansion module. Enables the administration of the buttons for the expansion module.
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Station: Page 1
EMU Login Allowed
Enables or disables using the station as a visited station by an Enterprise Mobility User (EMU).
Endpt ID
Available only if Endpt Init is enabled.
Valid Entry 00 to 62
Usage
A unique two-digit number for this endpoint. Each Endpt ID field must have a unique value for each endpoint on the same port.
Fixed TEI
Indicates whether or not the endpoint has a fixed Terminal Endpoint Identifier (TEI). The TEI identifies a unique access point within a service. TEIs must be administered for fixed TEI terminals.
Available only for ISDN-BRI data modules, NI-BRI telephones and WCBRI data modules.
Hunt-to Station
The extension the system must hunt to for this telephone when the telephone is busy. You can create a station hunting chain by assigning a hunt-to station to a series of telephones.
IP Softphone
Indicates whether or not this extension is either a PC-based multifunction station or part of a telecommuter complex with a call-back audio connection.
Available only for DCP station types and IP Telephones.
IP Video Softphone
Indicates whether or not this extension is a video softphone. Available only if IP Softphone is enabled.
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Station
IP Video
Use this field to specify whether the extension has IP video capability. The system displays this field for H.323 and SIP station types.
ITC (Information Transfer Capability)
The type of transmission facilities used for ISDN calls originating from this endpoint. Unavailable for voice-only or BRI stations.
Valid Entry restricted
unrestricted
Usage
Either restricted or unrestricted transmission facilities are used to complete the call. A restricted facility is a transmission facility that enforces 1's density digital transmission. In other words, a sequence of eight digital zeros are converted to a sequence of seven zeros and a digital one.
Only unrestricted transmission facilities are used to complete the call. An unrestricted facility is a transmission facility that does not enforce 1's density digital transmission. In other words, digital information is sent exactly as is.
Lock Messages
Controls access to voice messages by other users.
Valid Entry y
n
Usage
Restricts other users from reading or canceling the voice messages, or retrieving messages using Voice Message Retrieval.
Allows other users to read, cancel, or retrieve messages.
Location
The system displays this field only when you set the Multiple Locations field on the system parameters customer options screen to y, and set the Type field to H.323 or SIP station types.
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Valid entry 1 to 2000
blank
Station: Page 1
Usage (Depending on your server configuration, see Avaya Aura® Communication Manager System Capacities Table.) Assigns the location number to a particular station. Allows IP telephones and softphones connected through a VPN to be associated with the branch an employee is assigned to. This field is one way to associate a location with a station. For the other ways and for a list of features that use location, see the Location sections in Avaya Aura® Communication Manager Feature Description and Implementation. Indicates that the existing location algorithm applies. By default, the value is blank.
Loss Group
Valid Entry 1 to 17
Usage
Determines which administered two-party row in the loss plan applies to each station. Is not displayed for stations that do not use loss, such as x-mobile stations.
Mapping Mode
This field can be configured when the set type is XMOBILE. Controls the mode of operation in which the cell telephone operates when mapped to this XMOBILE extension. An XMOBILE station can be bridged to a deskset. These restrictions or modes exist because the COR of a bridge is ignored; instead the principal's COR is used. With Mapping Mode, an XMOBILE station can function as a bridge and still be restricted.
When a cell telephone is mapped to more than one XMOBILE station, then only one of the mapped XMOBILE stations can have origination or both as its Mapping Mode. Therefore, only one of the XMOBILE stations mapped to the cell telephone number is permitted to originate calls.
Valid Entry both none origination
termination
Usage
The cell telephone can be used to originate and terminate calls from its associated XMOBILE extension. This is the default when the XMOBILE type is PHS or DECT.
The XMOBILE station is disabled administratively and cannot originate and terminate calls from its associated internal extension.
The cell telephone can be used only to originate calls from its associated internal XMOBILE extension by dialing into the office server running Communication Manager.
The cell telephone can be used only to terminate calls from its associated internal XMOBILE extension. This is the default when the XMOBILE type is EC500.
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Station
Map-to Station
The extension of a physical telephone used for calls to a virtual extension. Cannot be used with an xmobile, xdid or any other virtual extension.
This field is applicable only for the virtual endpoints.
Message Lamp Ext
The extension of the station tracked with the message waiting lamp.
Media Complex Ext
When used with Multi-media Call Handling, indicates which extension is assigned to the data module of the multimedia complex. Users can dial this extension to place either a voice or a data call, and voice conversion, coverage, and forwarding apply as if the call were made to the 1number.
Valid Entry A valid BRI data extension H.323 station extension
blank
Usage
For MMCH, enter the extension of the data module that is part of this multimedia complex.
For 4600 series IP Telephones, enter the corresponding H.323 station. For IP Softphone, enter the corresponding H.323 station. If you enter a value in this field, you can register this station for either a road-warrior or telecommuter/Avaya IP Agent application.
Leave this field blank for single-connect IP applications.
Message Waiting Indicator
Specifies the type of message waiting indicator. This field is independent of the administration of the Caller ID Message Waiting Indication for CallrID telephones. Must be set to a value other than none when the station type is set to H.323.
Available only for ISDN-BRI data modules and for 500, 2500, K2500, 7104A, 6210, 6218, 6220, 8110, H.323 and VRU telephones.
Valid Entries led
Usage The message waiting indicator is a light-emitting diode (LED).
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Valid Entries neon
none
Station: Page 1
Usage The message waiting indicator is a neon indicator.
Note: The neon message waiting indicator is supported only on a small subset of boards, including older US-only boards, such as the TN746 and the TN793. Check the documentation for your board to see if neon is supported. Available only if the Analog Ringing Cadence is set to 1 (US) for the location parameters. No message waiting indicator is selected. This is the default.
MIM Mtce/Mgt
Indicates if the telephone supports MIM Maintenance and Management capabilities other than endpoint initialization. Available only if MIM Support is enabled.
MIM Support (Management Information Message Support)
Enables or disables MIM endpoint initialization (SPID support) and other Maintenance or Management capabilities.
Message Waiting Type
This field can be configured when the set type is XMOBILE.
Valid Entry DISPL
ICON NONE
Usage
The MW text is added to the end of the display line during an incoming or outgoing call.
The MWI message is used to update to the Wireless Terminal.
No MWI message is sent to the Wireless Terminal
Mute Button Enabled
Enables or disables the mute button on the station.
Mobility Trunk Group
Associates the XMOBILE station to a trunk.
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Station Valid Entries 2000 aar
ars
blank
Usage
A valid trunk group number for mobility routing. This trunk group is used for routing.
The Automatic Alternate Routing (AAR) capabilities of Communication Manager are used to direct the call to an ISDN trunk. If no ISDN trunk is available, the call is not extended out of the server. It provides ringback to the calling company and might eventually go to coverage.
The Automatic Route Selection (ARS) capabilities of Communication Manager are used to direct the call to an ISDN trunk. If no ISDN trunk is available, the call is not extended out of the server. It provides ringback to the calling company and might eventually go to coverage.
Not administered.
Model
The model of the NI-BRI telephone.
Valid Entry
Usage
L-3 Communication The NI-BRI telephone is a model L-3 Communication STE. STE
Tone Commander The NI-BRI telephone is a model 6210 and 6220 Tone Commander.
Other
The NI-BRI telephone is another model (for example, a Nortel 5317T).
Name
The name of the person associated with this telephone or data module. The system uses this value to create the system directory.
Note:
This field is supported by Unicode language display for the 4610SW,4620SW, 4621SW, and 4622SW telephones.
For more information on Unicode language display, see Administering Avaya Aura® Communication Manager.
Note:
In the display for emergency notification when completing the Name field, fill the most important identifying information at the beginning of the field. When an emergency call is made and a crisis alert station with a 27-character display is notified, only 17 characters of the Name field appear on the first display line, followed by the extension. The second line contains the last three characters of the Name field, followed by the word EMERGENCY. Characters 18 through 24 of the Name field are not displayed at all.
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Station: Page 1
Off Premises Station
Available only for analog telephones.
Valid Entries y
n
Usage
This telephone is located in a different building with the system. Requires administration of R Balance Network.
The telephone is located in the same building with the system.
Personalized Ringing Pattern
Defines the personalized ringing pattern for the station. Personalized Ringing allows users of some telephones to have one of 8 ringing patterns for incoming calls. For virtual stations, this field dictates the ringing pattern on its mapped-to physical telephone.
L = 530 Hz, M = 750 Hz, and H = 1060 Hz
Valid Entries 1 2 3 4 5 6 7 8
Usage MMM (standard ringing) HHH LLL LHH HHL HLL HLH LHL
Port
The Auxiliary and Analog ports assigned to the station are as follows.
Valid Entry 01 to 64 A to E 01 to 20 01 to 32
Usage The first and second numbers are the cabinet numbers. The third character is the carrier. The fourth and fifth characters are the slot numbers. G650 has 14 slots. The sixth and seventh characters are the port numbers.
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Station
Valid Entry x or X IP xxxVmpp
Analog Trunk port
Usage
Indicates that there is no hardware associated with the port assignment since the switch was set up, and the administrator expects that the extension has a non-IP set. Or, the extension had a non-IP set, and it dissociated. Use x for Administered WithOut Hardware (AWOH) and Computer Telephony (CTI) stations, as well as for SBS Extensions.
Indicates that there is no hardware associated with the port assignment since the switch was set up, and the administrator expects that the extension would have an IP set. This is automatically entered for certain IP station set types, but you can enter for a DCP set with softphone permissions. This changes to the s00000 type when the set registers.
Specifies the Branch Gateway.
· xxx is the Branch Gateway number, which is in the range 001 to 250.
· m is the module number, which is in the range 1 to 9.
· pp is the port number, which is in the range 01 to 32.
Analog trunk port is available with:
· MM711 and MM714 media modules
· TN747 and TN797 circuit packs
Passive Signalling Station
The entries y and n indicate the call processing status of the analog trunk port user. This field is available only when you select an analog trunk port as the station type.
Valid Entry y n
Usage
The analog station port does not process calls when off hook.
The analog station port processes and receives calls as an analog station. This is the default value.
R Balance Network
You must enter a value in this field if you set the Off-Premise Station field to y.
Valid Entry y
n
Usage
Selects the Resistor Only Balance network. The Resistor Only Balance network is used when Communication Manager is connected to terminal equipment, such as an SLC carrier or an impedance compensator, that provides an input impedance of 600 ohm and Communication Manager is less than 3,000 feet from the equipment.
Selects the Resistor and Capacitor (RC) balance network. RC balance network, when connected to off-premise stations, provide better signal transmission than Resistor only balance network.
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Station: Page 1
Remote Office Phone
Indicates whether or not this station is used as an endpoint in a remote office configuration.
Set
Appears only when the selected station type is CS1000.
Security Code
The security code required by users for specific system features and functions are as follows: · Extended User Administration of Redirected Calls · Personal Station Access · Redirection of Calls Coverage Off-Net · Leave Word Calling · Extended Call Forwarding · Station Lock · Voice Message Retrieval · Terminal Self-Administration · Enterprise Mobility User · Extension to Cellular · Call Forwarding · Posted Messages · Security Violation Notification · Demand Printing
The required security code length is administered system wide.
Short/Prefixed Registration Allowed
If the IP Stations or the IP Softphone field is set to y, the system displays the Short/Prefixed Registration Allowed field for this H.323 IP set types: 1603, 1608, 1616, 4601, 4601P, 4602, 4602P, 4606, 4610, 4612, 4620, 4622, 4624, 4630, 9610, 9620, 9630, and 9650.
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Station
Valid Entry y
n
default
Usage
Call processing allows the short registration to proceed per station basis. This entry overrides the System Parameters IP Options screen field value for that station.
Call processing rejects the short registration per station basis. This entry overrides the System Parameters IP Options screen field value for that station.
Call processing uses the Short/Prefixed Registration Allowed field option on the System Parameters IP Options screen for short registration. The default value is default.
Related links Short/Prefixed Registration Allowed on page 477
Speakerphone
Controls the behavior of speakerphones.
Valid Entry 1-way 2-way grp-listen
none
Usage
Indicates that the speakerphone listen-only.
Indicates that the speakerphone is both talk and listen.
With Group Listen, a telephone user can talk and listen to another party with the handset or headset while the telephone's two-way speakerphone is in the listen-only mode. Others in the room can listen, but cannot speak to the other party through the speakerphone. The person talking on the handset acts as the spokesperson for the group. Group Listen provides reduced background noise and improves clarity during a conference call when a group needs to discuss what is being communicated to another party.
Available only with 6400-series and 2420/2410 telephones.
Not administered for a speakerphone.
Signaling Type
The Signaling Type field is used to specify how the A and B bits are used on an Off Premise Station (OPS) line.
Valid Entry Fx
Inverted
Usage
The signaling bits are manipulated as if the port is terminating on a Foreign Exchange (FX) unit in a D4 channel bank. This value is default.
The signaling bits are manipulated as if the port is terminating on a Special Access Unit (SAU) in a D4 channel bank.
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Station: Page 1
Survivable COR
Sets a level of restriction for stations to be used with the survivable dial plan to limit certain users to only to certain types of calls. You can list the restriction levels in order from the most restrictive to least restrictive. Each level has the calling ability of the ones above it.
Available for all analog and IP station types.
Valid Entries emergency internal local
toll
unrestricted
Usage
This station can only be used to place emergency calls.
This station can only make intra-switch calls. This is the default.
This station can only make calls that are defined as locl, op, svc, or hnpa in the Survivable Gateway Call Controller's routing tables.
This station can place any national toll calls that are defined as fnpa or natl on the Survivable Gateway Call Controller's routing tables.
This station can place a call to any number defined in the Survivable Gateway Call Controller's routing tables. Those strings marked as deny are also denied to these users.
Survivable Trunk Dest
Designates certain telephones as not being allowed to receive incoming trunk calls when the Branch Gateway is in survivable mode.
Available for all analog and IP station types.
Valid Entry y
n
Usage
Allows this station to be an incoming trunk destination while the Branch Gateway is running in survivability mode. This is the default.
Prevents this station from receiving incoming trunk calls when in survivable mode.
Survivable GK Node Name
Any valid previously-administered IP node name. Identifies the existence of other H.323 gatekeepers located within gateway products that offer survivable call features. For example, the MultiTech MVPxxx-AV H.323 gateway family and the SLS function within the Branch Gateways. When a valid IP node name is entered into this field, Communication Manager adds the IP address of this gateway to the bottom of the Alternate Gatekeeper List for this IP network region. As H.323 IP stations register with Communication Manager, this list is sent down in the registration confirm message. With this, the IP station can use the IP address of this Survivable Gatekeeper as the call controller of last resort.
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Station
If blank, there are no external gatekeeper nodes within a customer's network. This is the default value. Available only if the station type is an H.323 station for the 46xx or 96xx models.
Secure Terminal Equip
Used when Secure Terminal Equipment (STE) telephones are administered as 8510 telephones. Available only for 8503, 8510, and 8520 stations in Communication Manager 3.0 and later.
Valid Entries y
n
Usage
Leave the Bearer Cap IE unchanged. 3.1khz is used to let secure calls from Secure Terminal Equipment (STE) telephones work properly.
Force the Bearer Cap IE to "speech" before a call is delivered to the 85xx BRI station.
SPID -- (Service Profile Identifier)
The Service Profile IDentifier (SPID) for this station. The SPID is a variable length parameter required for ISDN-BRI stations.
The SPID is a numeric string, which means that the value of 00 is different from 000. The SPID must be different for all terminals on the BRI and from the Service SPID. The SPID should always be assigned. If the SPID is not assigned for the first BRI on a port, any other BRI assignment to that port is blocked.
Note:
If you have not administered a port for an ISDN-BRI extension and intend to use Terminal Translation Initialization (TTI) to assign the port, then the SPID number must equal the station number.
Available only if the terminal supports ISDN terminal initialization procedures.
Special Dialing Option
Identifies the type of dialing for calls when this data module originates calls.
Valid Entry hot-line
default blank
Usage
The Hot Line Service destination number is stored in an Abbreviated Dialing List. When the user goes off-hook on a Data Hot Line call, the system dials the AD number.
The system dials the default dialing AD number.
Normal keyboard dialing.
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Station: Page 1
System ID
This field can be configured for CS 1000 station types only. This field allows you to leave the field blank or enter a string of up to 9 characters. With Release 8.0 and later, more than one station can use the combination of System ID and Terminal Number.
With Release 8.0.1 and later, each station must have a unique combination of System ID and Terminal Number.
TEI
Available only if the station has a fixed Terminal Endpoint Identifier (TEI).
Valid Entry 0 to 63
Usage A one- or two-digit number that identifies the TEI of the station.
Tests
Valid Entry y n
Usage
Enables port maintenance tests.
Disables port maintenance tests. Required if the equipment (dictaphone) connected to the port does not support these tests.
Time of Day Lock Table
Valid Entry 1 to 5
blank
Usage
Assigns the station to a Time of Day (TOD) Lock/Unlock table. The assigned table must be administered and active.
Indicates no TOD Lock/Unlock feature is active. This is the default.
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
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Station
Type
Use this field to specify the telephone type. You must administer the station type for each station that you add to the system.
The following table lists the telephones, virtual telephones, and personal computers that you can administer on Communication Manager. Telephones that are not in the table require an alias to a supported set type.
Note:
If Terminal Translation Initialization is enabled, you cannot change the analog telephones administered with hardware to a virtual extension.
Telephone type Single-line analog
CallerID
Single-line DS1/DSO Terminals
Model 500 2500, 2500 with Message Waiting Adjunct 6210 6211 6218 6219 6220 6221 Analog telephone with Caller ID 7101A, 7102A 7103A Programmable and Original 7104A 8110 DS1FD 7302H, 7303H Voice Response Unit (VRU) with C&D tones VRU without C&D tones DS1 device without forward disconnect VRU with forward disconnect without C&D tones VRU with forward disconnect without C&D tones 510D
Administered as 500 2500
6210 6210 6218 6218 6220 6220 CallrID 7101A 7103A 7104A 8110 DS1FD 7303S VRU
2500 ops ds1fd or ds1sa
VRUFD or VRUSA
510
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Telephone type Multi-appearance hybrid
Multi-appearance digital
Model 515BCT 7303S 7305H 7305S 7309H 7313H 7314H 7315H 7316H 7317H 2402 2410 2420 6402 6402D 6408 6408+ 6408D 6408D+ 6416D+ 6424D+ 7401D 7401+ 7403D 7404D 7405D 7406D 7406+ 7407D 7407+ 7410D 7410+ 7434D 7444D 8403B
Station: Page 1
Administered as 515 7303S, 7313H 7305S 7305S, 7316H, 7317H 7309H, 7313H 7313H 7314H 7315H 7316H 7317H 2402 2410 2420 6402 6402D 6408 6408+ 6408D 6408D+ 6416D+ 6424D+ 7401D 7401+ 7403D 7404D 7405D 7406D 7406+ 7407D 7407+ 7410D 7410+ 7434D 7444D 8403B
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Station
Telephone type IP Telephone
Model 8405B 8405B+ 8405D 8405D+ 8410B 8410D 8411B 8411D 8434D 9404 9408 CALLMASTER I CALLMASTER II, III, IV CALLMASTER VI IDT1 IDT2 4601+
Note:
When you add a new 4601 IP telephone, you must use the 4601+ station type. This station type has the Automatic Callback feature enabled. 4602+
Note:
When you add a new 4602 IP telephone, you must use the 4602+ station type. This station type has the Automatic Callback feature enabled. 4606 4610 4612 4620SW IP (G3.5 hardware) 4621 4622 4624
Administered as 8405B 8405B+ 8405D 8405D+ 8410B 8410D 8411B 8411D 8434D 9404 9408 602A1 603A1, 603D1, 603E1, 603F1 606A1 7403D 7406D 4601+
4602+
4606 4610 4612 4620 4621 4622 4624
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Station: Page 1
Telephone type SIP IP Telephone
Model 4625 4690 9608 9610 9611 9620 9621 9630 9640 9641 9650 · 4602SIP with SIP firmware
· 4610SIP with SIP firmware
· 4620SIP with SIP firmware
· 4620SIP CC (Call Center)
· SIP Softphone/Avaya one-X Desktop
· Toshiba SP-1020A
Note:
You must administer any telephone with SIP firmware used for SIP networking as a 4620SIP, 96xxSIP, or 16CC SIP telephone.
Note:
Communication Manager Release 6.2 and later do not support 1616SIP CC and 4620SIP CC telephones. 9601SIP 9620, 9630, 9630G 9640, 9640G with SIP firmware 9608 with SIP firmware 9611 with SIP firmware 9621 with SIP firmware 9641 with SIP firmware 9608 with SIP firmware (for call center) 9611 with SIP firmware (for call center)
Administered as 4625 4690 9608 9610 9611 9620 9621 9630 9640 9641 9650 4620SIP
9608SIP 96xx or 96xxSIP telephone
9608SIP 9611SIP 9621SIP 9641SIP 9608SIPCC 9611SIPCC
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Station
Telephone type
H.323 SoftPhone
Personal computer (voice/data) Test Line No hardware assigned at the time of administration.
Model 9621 with SIP firmware (for call center) 9641 with SIP firmware (for call center) application Road Warrior Native H.323 Singleconnect Any NI-BRI (N1 and N2) telephone 7505D 7506D 7507D 8503D 8510T 8520T 6300/7300 6538/9 ATMS
Key telephone system interface AWOH
CTI XMOBILE ISDN-BRI data module SBS Extension
--
any digital set CTI station CTI station EC500, DECT, PHS 7500
SBS test extension (no hardware)
Administered as 9621SIPCC 9641SIPCC H.323 or DCP type H.323 H.323 or DCP type NI-BRI 7505D 7506D 7507D 8503D 8510T 8520T PC Constellation 105TL · Use XDID when Communication
Manager assigns a DID number to the station. · Use XDIDVIP when the administrator later assigns a DID number to the station. · Use virtual to map the station and other extensions to one physical telephone. K2500
-- CTI CTI XMOBILE 7500
sbs
Terminal Number
This field can be configured for CS 1000 station types only. You can enter numbers in the following range: 0.0.0.0 to 252.1.15.31.
The first digit must be divisible by 4. For example: 0, 4, 8, ..., 252.
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Station: Page 1
Unicode Name
This read-only field shows the assignment of Unicode Name to the station.
Valid entry n y
Usage Unicode Name is not assigned to the station. Unicode Name is assigned to the station.
For more information on Unicode Language Display, see Administering Avaya Aura® Communication Manager.
XOIP Endpoint type
Available only for 500, 2500, K2500 and CallrID station types.
Valid Entries auto modem fax tty
Usage
Identifies the endpoint type of the analog station. Use this field for older or nonstandard external equipment such as modems, fax, and TTY devices that are not easily recognized by VoIP resources within Communication Manager. By identifying this external equipment through administration, VoIP firmware can determine whether to immediately attempt to put a call in pass-through mode, or allow the system to handle it normally. Default is auto.
Note:
This field is intended for exception cases only. For the majority of stations, use the default setting of auto.
XID
Use this field to enable the Layer 2 XID testing capability. The system displays this field only for an ISDN-BRI data module.
XMOBILE Type
The type of XMOBILE station. Available only if the Type field is XMOBILE and the Mobility Trunk Group field is administered.
Valid Entry DECT
Usage
The DECT Access System or the AGCS (ROAMEO) IS-136 (TDMA cellular). This represent the ISDN-based DECT system.
Table continues...
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Station
Valid Entry EC500 IPDECT PHS
Usage
Any public cellular networks.
The H.323 IP-based DECT system. This can either be the ASCOM or the DeTeWe product.
The DENSO 300M.
Station: Page 2
Att. Call Waiting Indication
With attendant call waiting, attendant-originated or attendant-extended calls to a busy single-line telephone can wait and sends distinctive call-waiting tone to the single-line user.
Valid Entry y n
c
Usage Activates Call Waiting for the telephone without Caller ID information. This feature must be enabled when the Type is set to H.323. This is the default. Call Waiting is disabled for the station.
Disable this feature if:
· Data Restriction is enabled
· Switchhook Flash field is disabled
· Data Privacy is enabled for the telephone's class of service (COS) Enables the Caller ID Delivery with Call Waiting feature, which displays CID information on for the waiting call. Available only if the station Type is CallrID.
Active Station Ringing
Defines how calls ring to the telephone when it is off-hook without affecting how calls ring at this telephone when the telephone is on-hook.
Valid Entry continuous single if-busy-single
silent
Usage
All calls to this telephone ring continuously.
Calls to this telephone receive one ring cycle and then ring silently.
Calls to this telephone ring continuously when the telephone is off-hook and idle. Calls to this telephone receive one ring cycle and then ring silently when the telephone is off-hook and active.
All calls to this station ring silently.
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Station: Page 2
AUDIX Name
The voice messaging system associated with the station. Must contain a user-defined adjunct name that was previously administered.
Always Use
Use this field to enable the following emergency call handling settings:
· A softphone can register irrespective of the emergency call handling settings the user has entered into the softphone. If a softphone dials 911, the value administered in the Emergency Location Extension field is used as the calling party number. The user-entered emergency call handling settings of the softphone are ignored.
· If an IP telephone dials 911, the value administered in the Emergency Location Extension field is used as the calling party number.
· If an agent dials 911, the physical station extension is used as the calling party number, overriding the value administered in the LoginID for ISDN Display field.
Does not apply to SCCAN wireless telephones, or to extensions administered as type H.323.
Auto Answer
In an Expert Agent Environment (EAS) environment, the auto answer setting for an Agent LoginID overrides the endpoint settings when the agent logs in.In EAS environments, the auto answer setting for the Agent LoginID can override a station's setting when an agent logs in.
Valid entry all
acd
none icom
Usage
All ACD and non-ACD calls to an idle station cut through immediately. The agent cannot use automatic hands-free answer for intercom calls. With non-ACD calls, the station rings while the call is cut through. To prevent the station from ringing, activate the ringer-off feature button, provided the Allow Ringer-off with Auto-Answer feature is enabled for the system.
Only ACD split, ACD skill, and direct agent calls cut through. Non-ACD calls to the station ring audibly.
For analog stations:
· Only the ACD split or skill calls and direct agent calls cut through.
· Non-ACD calls receive busy treatment. If the station is active on an ACD call and a non-ACD call arrives, the agent receives call-waiting tone.
All calls to the station receive an audible ringing.
The user can answer an intercom call from the same intercom group without pressing the intercom button.
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Audible Message Waiting
Audible Message Waiting field enables or disables an audible message waiting tone indicating the user has a waiting message consisting of a stutter dial tone when the user goes off-hook.
This field does not control the Message Waiting lamp.
Available only if Audible Message Waiting is enabled for the system.
Automatic Moves
With Automatic Moves, you can unplug a DCP telephone from one location and moved to a new location without additional Communication Manager administration. Communication Manager automatically associates the extension to the new port.
Caution:
When a DCP telephone is unplugged and moved to another physical location, the Emergency Location Extension must be changed for that extension or the USA Automatic Location Identification data base must be manually updated. If the Emergency Location Extension is not changed or if the USA Automatic Location Identification data base is not updated, the DID number sent to the Public Safety Network could send emergency response personnel to the wrong location.
Valid Entry always once
no done error
Usage
The DCP telephone can be moved anytime without additional administration by unplugging from one location and plugging into a new location.
The DCP telephone can be unplugged and plugged into a new location once. After a move, the field is set to done the next time that routine maintenance runs on the DCP telephone.
Use when moving a large number of DCP telephones so each extension is removed from the move list. Also use to prevent automatic maintenance replacement.
Requires administration to move the DCP telephone.
Communication Manager sets the field to done after the telephone is moved and routine maintenance runs on the DCP telephone.
Communication Manager sets the field to error, after routine maintenance runs on the DCP telephone, when a non-serialized telephone is set as a movable telephone.
Auto Select Any Idle Appearance
Auto Select Any Idle Appearance field enables or disables automatic selection of any idle appearance for transferred or conferenced calls. Communication Manager first attempts to find an
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idle appearance that has the same extension number as the call being transferred or conferenced has. If that attempt fails, Communication Manager selects the first idle appearance.
Adjunct Supervision
Available only if the station type is 500, 2500, k2500, 8110, ops, ds1fd, ds1sa, VRU, VRUFD, or VRUSA.
Valid Entry y
n
Usage
An analog disconnect signal is sent automatically to the port after a call terminates. Analog devices, such as answering machines and speakerphones, use this signal to turn the devices off after a call terminates.
Required hunt group agents are alerted to incoming calls. In a hunt group environment, the disconnect signal blocks the reception of zip tone and incoming call notification by an auto-answer station when a call is queued for the station.
Busy Auto Callback without Flash
Enables or disables automatic callback for a calling analog station without flashing the hook. Available for analog telephones only if Without Flash is enabled for the system.
Bridged Call Alerting
Controls how the user is alerted to incoming calls on a bridged appearance.
Valid Entry y n
Usage
The bridged appearance rings when a call arrives at the primary telephone.
The bridged appearance flashes but does not ring when a call arrives at the primary telephone. This is the default.
If disabled and Per Button Ring Control is also disabled, audible ringing is suppressed for incoming calls on bridged appearances of another telephone's primary extension.
Bridged Idle Line Preference
Use this field to specify that the line that the system selects when you go off hook is always an idle call appearance for incoming bridged calls.
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Station
Valid entry y n
Usage The user connects to an idle call appearance instead of the ringing call. The user connects to the ringing bridged appearance.
CDR Privacy
Enables or disables Call Privacy for each station. With CDR Privacy, digits in the called number field of an outgoing call record can be blanked on a per-station basis. The number of blocked digits is administered system-wide as CDR parameters.
Coverage Msg Retrieval
Allows or denies users in the telephone's Coverage Path to retrieve Leave Word Calling (LWC) messages for this telephone. Applies only if the telephone is enabled for LWC Reception.
Coverage After Forwarding
Governs whether an unanswered forwarded call is provided coverage treatment.
Valid Entry y
n
s(ystem)
Usage
Coverage treatment is provided after forwarding regardless of the administered system-wide coverage parameters.
No coverage treatment is provided after forwarding regardless of the administered system-wide coverage parameters.
Administered system-wide coverage parameters determine treatment.
Call Waiting Indication
User, attendant-originated, and outside calls to a busy single-line telephone can wait and sends a distinctive call-waiting tone to the single-line user. Not available if Data Restriction is enabled, Switchhook Flash is disabled, or if Data Privacy is active by way of the telephone COS assignment.
Valid Entry y
n c
Usage
Activates Call Waiting without Caller ID information for the telephone. This is the default.
Call Waiting is disabled for the station.
Enables the Caller ID Delivery with Call Waiting feature, which displays CID information on for the waiting call. This value can only be entered when the station type is CallrID.
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Call Appearance Display Format
Specifies the display format for the station. Bridged call appearances are not affected by this field. This field is available only on telephones that support downloadable call appearance buttons, such as the 2420 and 4620 telephones.
Note: This field sets the administered display value only for an individual station.
Valid Entry loc-param-default inter-location
intra-location
Usage
The system uses the administered system-wide default value. This is the default.
The system displays the complete extension on downloadable call appearance buttons.
The system displays a shortened or abbreviated version of the extension on downloadable call appearance buttons.
Conf/Trans On Primary Appearance
Enables or disables the forced use of a primary appearance when the held call to be conferenced or transferred is a bridge. This is regardless of the administered value for Auto Select Any Idle Appearance.
Data Restriction
Enables or disables data restriction that is used to prevent tones, such as call-waiting tones, from interrupting data calls. Data restriction provides permanent protection and cannot be changed by the telephone user. Cannot be assigned if Auto Answer is administered as all or acd. If enabled, whisper page to this station is denied.
Display Client Redirection
Enables or disables the display of redirection information for a call originating from a station with Client Room Class of Service and terminating to this station. When disabled, only the client name and extension or room display. Available only if Hospitality is enabled for the system.
Note:
This field must be enabled for stations administered for any type of voice messaging that needs display information.
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Direct IP-IP Audio Connections
Use this field to enable direct audio connections between IP endpoints. Direct audio connections save bandwidth resources and improve the sound quality of voice over IP transmissions.
Note:
This field has no impact on a SIP station (OPS) on shuffling. For SIP stations, this field is controlled by the "Direct IP-IP Audio Connections" field in the Signaling group form.
Distinctive Audible Alert
Use this field to enable the distinctive audible alerts to allow telephones to receive three different types of ringing patterns that identify the type of incoming calls. Distinctive ringing might not work properly for off-premises telephones.
Emergency Location Ext
The Emergency Location Extension for this station. This extension identifies the street address or nearby location when an emergency call is made. Defaults to the telephone's extension. Accepts up to sixteen digits.
Note:
On the ARS Digit Analysis Table in Communication Manager, 911 must be administered to be call type emer or alrt for the E911 Emergency feature to work properly.
Related links Remote Soft Phone Emergency Calls on page 774
EC500 State
Use this field to observe or change the EC500 state of the station .This status is also observed and changed by administering an EC500 button on the station.
Valid Entry enabled
disabled
Usage
When EC500 state is enabled, a call to that station generates a new outbound call from the Communication Manager to the configured EC500 destination, typically a mobile phone.
When EC500 state is disabled, the outbound call to the EC500 destination is not generated.
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H.320 Conversion
Use this field to enable the conversion of H.320-compliant calls to voice-only calls for the attendant console.
Note:
The system can handle only a limited number of conversion calls. Therefore, the number of attendant consoles with H.320 conversion must be limited.
Idle/Active Ringing (Callmaster)
Defines how a call rings to the telephone when it is on-hook. Applies to CALLMASTER telephones. This field is available only if station type is 602A1.
Valid Entry continuous if-busy-single
silent-if-busy single
Usage
All calls to this telephone ring continuously.
Calls to this telephone ring continuously when the telephone is off-hook and idle, and calls to this telephone receive one ring cycle and then ring silently when the telephone is off-hook and active.
Calls ring silently when this station is busy.
Calls to this telephone receive one ring cycle and then ring silently.
IP Audio Hairpinning
Indicates whether or not IP endpoints are allowed to connect through the IP circuit pack in the server without going through the time division multiplexing (TDM) bus.
Available only if the station type is one of the several h.323 or sip types.
Ignore Rotary Digits
Indicates whether or not rotary digits from the set are ignored. If enabled, the short switch-hook flash (50 to 150) from a 2500-type set is ignored.
LWC Reception
Use this field to specify the location where the system must store the LWC messages.
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Station
Valid entry spe none audix
Usage Use this option to store the LWC messages on Switch Processor Element (SPE). Use this option if you do not want to store the LWC messages. Use this option to store the LWC messages on the voice messaging system.
Related links AUDIX Name on page 411
LWC Activation
Activates or deactivates the Leave Word Calling (LWC) feature. With LWC, internal telephone users on this extension can leave short pre-programmed messages for other internal users.
You must use LWC if:
· The system has hospitality and the guest-room telephones require LWC messages indicating that wakeup calls failed
· The LWC messages are stored in a voice-messaging system
LWC Log External Calls
This field can be configured when the set type is XMOBILE. Determines whether or not unanswered external call logs are available to end users. When external calls are not answered, Communication Manager keeps a record of up to 15 calls provided information on the caller identification is available. Each record consists of the latest call attempt date and time.
Multimedia Early Answer
Enables or disables multimedia early answer on a station-by-station basis.
You must enable the station for the Multimedia Early Answer feature if the station receives coverage calls for multimedia complexes, but is not multimedia-capable. This ensures that calls are converted and the talk path is established before ringing at this station.
MWI Served User Type
Controls the auditing or interrogation of a served user's message waiting indicator (MWI).
Valid Entries fp-mwi
Usage The station is a served user of an fp-mwi message center.
Table continues...
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Valid Entries qsig-mwi sip-adjunct blank
Station: Page 2
Usage The station is a served user of a qsig-mwi message center. Used to audit message waiting lamps. The served user's MWI is not audited or if the user is not a served user of either an fp-mwi or qsig-mwi message center.
PCOL/TEG Call Alerting
Enables or disables alerting at the station for Personal CO Line/Terminating Extension Group calls.
Available only for 510 telephones.
Per Button Ring Control
Using this option you can enable or disable ring control for every button, provided you have the station user credentials.
Valid Entries y
n
Usage To enable Automatic Abbreviated and Delayed ring transition for each call-appr on the station, select ring behavior individually for each call-appr or brdg-appr option.
To prevent the system from automatically moving the line selection to a silently alerting call, unless the call was audibly ringing earlier.
Note:
The abrdg-appr option is unavailable for SIP station. To enable the calls on call-appr buttons always to ring the station
To enable the calls on brdg-appr buttons always ring or not ring based on the Bridged Call Alerting value
To move line selection to a silently alerting call, if the call is not audibly ringing the station
Note:
The abrdg-appr option is unavailable for SIP station.
Per Station CPN - Send Calling Number
Determines Calling Party Number (CPN) information sent on outgoing calls from this station.
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Station
Valid Entries y n r blank
Usage
All outgoing calls from the station deliver the CPN information as "Presentation Allowed."
No CPN information is sent for the call.
Outgoing non-DCS network calls from the station delivers the Calling Party Number information as "Presentation Restricted."
The sending of CPN information for calls is controlled by administration on the outgoing trunk group the calls are carried on.
Precedence Call Waiting
Activates or deactivates Precedence Call Waiting for this station.
Redirect Notification
Enables or disables redirection notification that gives a half ring at this telephone when calls to this extension are redirected through Call Forwarding or Call Coverage. Must be enabled if LWC messages are stored on a voice-messaging system.
Restrict Last Appearance
Valid Entries y
n
Usage
Restricts the last idle call appearance used for incoming priority calls and outgoing call originations only.
Last idle call appearance is used for incoming priority calls and outgoing call originations.
Recall Rotary Digit
Enables or disables the Recall Rotary Digit dialing that enables this user to perform conference and transfer operations.
Available only if the station type is 500 or 2500.
Remote Soft Phone Emergency Calls
Tells Communication Manager how to handle emergency calls from the IP telephone.
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Caution:
An Avaya IP endpoint can dial emergency calls (for example, 911 calls in the U.S.). It only reaches the local emergency service in the Public Safety Answering Point area where the telephone system has local trunks. You cannot use an Avaya IP endpoint to dial to and connect with local emergency service when dialing from remote locations that do not have local trunks. Avoid using an Avaya IP endpoint to dial emergency numbers for emergency services when dialing from remote locations. Avaya Inc. is not responsible or liable for any damages resulting from misplaced emergency calls made from an Avaya endpoint. Your use of this product indicates that you have read this advisory and agree to use an alternative telephone to dial all emergency calls from remote locations. If you have questions about emergency calls from IP telephones, go to the Avaya Support website at http:// support.avaya.com.
Available only if the station is an IP Softphone or a remote office station.
Valid Entry as-on-local
block cesid
Usage
If the emergency location extension that corresponds to this station's IP address is not administered (left blank), the value as-on-local sends the station emergency location extension to the Public Safety Answering Point (PSAP).
If the administrator populates the IP address mapping with emergency numbers, the value as-on-local functions as follows:
· If the station emergency location extension is the same as the IP address mapping emergency location extension, the value as-on-local sends the station's own extension to the Public Safety Answering Point (PSAP).
· If the station emergency location extension is different from the IP address mapping emergency location extension, the value as-on-local sends the IP address mapping extension to the Public Safety Answering Point (PSAP).
Prevents the completion of emergency calls. Use this entry for users who move around but always have a circuit-switched telephone nearby, and for users who are farther away from the server than an adjacent area code served by the same 911 Tandem office. When users attempt to dial an emergency call from an IP Telephone and the call is blocked, they can dial 911 from a nearby circuit-switched telephone instead.
Allows Communication Manager to send the CESID information supplied by the IP Softphone to the PSAP. The end user enters the emergency information into the IP Softphone.
Use this entry for IP Softphones with road warrior service that are near enough to the server that an emergency call reaches the PSAP that covers the softphone's physical location. If the server uses ISDN trunks for emergency calls, the digit string is the telephone number, provided that the number is a local direct-dial number with the local area code, at the physical location of the IP Softphone. If the server uses CAMA trunks for emergency calls, the end user enters a specific digit string for each IP Softphone location, based on advice from the local emergency response personnel.
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Valid Entry option
Usage
Allows the user to select the option (extension, block, or cesid) that the user selected during registration and the IP Softphone reported. This entry is used for extensions that can be swapped back and forth between IP Softphones and a telephone with a fixed location.
The user chooses between block and cesid on the softphone. A DCP or IP telephone in the office automatically selects the extension.
Service Link Mode
Use this field to specify the duration of a service link connection. The service link is the combined hardware and software multimedia connection between an H.320 Desktop Video Conferencing (DVC) system and Communication Manager.
The service link is established when a user receives or makes a call during a multimedia, IP softphone, or IP telephone session.
Valid entry asneeded
perman ent
Usage
For multimedia, IP softphone, and IP telephone users. The service link remains connected for 10 seconds after the user disconnects a call so that the user can immediately make or receive another call. After 10 seconds, the link is disconnected, and a new link must be established to make or receive a call.
For call center agents who are constantly making or receiving calls during the multimedia, IP softphone, or IP telephone session. The service link remains connected for the entire duration of the session.
Switchhook Flash
Valid Entry y
n
Usage
Allows users to use the switchhook flash function to activate Conference/Transfer/ Hold and Call Waiting. Required for H.323 station types.
Disables the flash function so that when the switchhook is pressed while active on a call, the call drops. Requires that Call Waiting Indication is disabled.
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Select Last Used Appearance
Valid Entry y
n
Usage
Indicates a station's line selection is not to be moved from the currently selected line button to a different, non-alerting line button. The line selection on an on-hook station only moves from the last used line button to a line button with an audibly alerting call. If there are no alerting calls, the line selection remains on the button last used for a call.
The line selection on an on-hook station with no alerting calls can be moved to a different line button that might be serving a different extension.
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Access Code
A five-digit access code used to place a wireless terminal into service. The access code is a temporary, shorter version of the complete User Authentication Key (UAK) required by the system when the terminal is first put into service. It is used to automatically generate a unique UAK for that wireless terminal over-the-air.
Available only if a wireless terminal model number is selected as the station type.
Bridged Appearance Origination Restriction
Restricts or allows call origination on the bridged appearance.
Valid Entry y n
Usage
Call origination on the bridged appearance is restricted.
Call origination ion the bridged appearance is allowed. This is normal behavior, and is the default.
Bridging Tone for This Extension
Allows you to enable or disable a single burst of tone when a station bridges on to the principal's call.
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Valid Entry y n
Usage Enables a single burst of tone when a station bridges on to the principal's call. Disables a single burst of tone when a station bridges on to the principal's call.
This is the default value for a new SIP station, or when a SIP station upgrades to Communication Manager 8.0 or later from a previous release.
Do not Share Port with Unnamed Registration
Controls whether x-port station can be shared or not when a request comes from an un-named IP address. This field is applicable for DCP phones with x-port only. This field appears only when Unnamed Registrations and PSA for IP Telephones field is set to y on the system-parameters features screen and DCP station.
Valid Entry y
n
Usage
Communication Manager does not allow x-port station to be shared when a request comes from un-named IP address.
Communication Manager allows x-port station to be shared when a request comes from un-named IP address. This is the default value.
Enhanced Callr-Info display for 1-line display phones
Use this field to enable or disable the Callr-Info Persists feature. Use this feature with telephones that have a one-line display and a callr-info button that displays the collected digits. For more information on Callr-info Persists, see Avaya Aura® Call Center Elite Feature Reference.
Valid entry n
Usage
This entry is the default. If an agent presses the callr-info button, the telephone displays the callrinfo digits. The callr-info digits are cleared after:
· the timeout period, defined by the Callr-info Display Timer field, expires or
· the agent disconnects the call
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Valid entry y
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Usage On receiving an incoming call with callr-info digits associated, the indicator of the callr-info button flashes.
· If the agent presses the callr-info button, the caller information is displayed on the telephone and the status lamp of the button becomes steady. If the agent presses the callr-info button after the telephone displays the caller information, the telephone displays the caller information again and the indicator of the callr-info button winks.
· If the agent does not press the button, the status lamp continues to flash during the call.
Forwarded Destination
A destination extension for both internal and external calls for each of the three types of enhanced call forwarding (Unconditional, Busy, and No Reply). Accepts up to 18 digits. The first digit can be an asterisk *.
Requires administration to indicate whether the specific destination is active (enabled) or inactive (disabled).
Home
Indicates the roaming status of the wireless user. Available only when a wireless terminal model number is selected as the station type.
Valid Entry y n
Usage The user's home. This is the default. The roaming system.
IP Phone Group ID
Available for H.323 and SIP station types.
Valid entry 0 to 999 blank
Usage The Group ID number for the station.
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IPEI
Available when a wireless terminal model number is selected as the station type.
Valid Entry 0 to 9 a to f A to F blank
Usage
The International Portable Equipment Identifier of the wireless terminal. Accepts a unique nine-character hexadecimal ID number.
Offline Call Logging
Use the Offline Call Logging field to back up and restore call logs for an offline user.
Valid entry y
n
Usage
Communication Manager tracks the calls that the user missed when the telephone was in the logged-out state, and updates the call logs on the telephone when the user logs in.
The default option is y.
Communication Manager does not track the calls that the user missed when the telephone was in the logged-out state.
Require Mutual Authentication if TLS
The TLS server, Communication Manager, and the TLS client, 96x1 H.323 phone, must both exchange Identity certificates and mutually authenticate the other side certificate.
If an H.323 phone has a certificate and the Require Mutual Authentication if TLS field is set to:
Valid entry y
n
Usage
Communication Manager validates the certificate. If the certificate is valid, Communication Manager allows TLS setup with H.323 phones. If the certificate is invalid, Communication Manager rejects the TLS request.
Communication Manager does not check for the phone certificate. Instead Communication Manager checks for the root certificate. If the root certificate is valid, Communication Manager allows TLS setup with H.323 phones to proceed without an endpoint certificate. If the root certificate is invalid or expired, Communication Manager rejects the TLS request.
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If an H.323 phone does not have a certificate and the Require Mutual Authentication if TLS field is set to:
Valid entry y
n
Usage
Communication Manager does not accept the security profile, H323TLS, in the Gatekeeper Confirmation (GCF). If the endpoint attempts to set up TLS, Communication Manager rejects the TLS request.
Communication Manager does not check for the phone certificate. Instead, Communication Manager checks for the root certificate. If the root certificate is valid, Communication Manager allows TLS setup with H.323 phones to proceed without an endpoint certificate. If the root certificate is invalid or expired, Communication Manager rejects the TLS request.
SAC/CF Override
With SAC/CF Override, the user of the calling station can override the redirection set by the called station.
Valid entry ask
no
yes
Usage
The system prompts the user of the calling station whether the call must follow the redirection path or override the redirection path. The user can type y or n.
The user of the calling station cannot override the redirection path of the call. The call follows the redirection path.
The user of the calling station can override the redirection path of the call, provided the called station has at least one idle call appearance.
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Auto-A/D
Enables or disables automatic abbreviated/delayed ringing for a call appearance. Available only if Per Button Ring Control is enabled.
Abbreviated Dialing List 1, List 2, List 3
Assigns up to three abbreviated dialing lists to each telephone.
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Station
Valid Entry enhanced group personal system
Usage
Telephone user can access the enhanced system abbreviated dialing list.
Telephone user can access the specified group abbreviated dialing list. Requires administration of a group number.
Telephone user can access and program their personal abbreviated dialing list. Requires administration of a personal list number.
Telephone user can access the system abbreviated dialing list.
Building
A valid building location.
Cable
Identifies the cable that connects the telephone jack to the system.
Cord Length
The length of the cord attached to the receiver. This is a free-form entry, and can be in any measurement units.
Floor
A valid floor location.
Headset
Indicates whether or not the telephone has a headset.
HOT LINE DESTINATION -- Dial Code
The dial code in the specified abbreviated dialing list where the hotline destination number is stored.
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HOT LINE DESTINATION -- Abbreviated Dialing Dial Code
Available only if Special Dialing Option is hot-line.
Hot Line Service is used when very fast service is required and when a telephone is used only for accessing a certain facility.
HOT LINE DESTINATION -- Abbreviated Dialing List Number
The abbreviated dialing list where the hotline destination number is stored.
Jack
Alpha-numeric identification of the jack used for this station.
Mounting
Indicates whether the station mounting is d(esk) or w(all).
Rg
Valid Entry a(bbreviated-ring) d(elayed-ring) n(o-ring) r(ing)
Usage
The type of automatic abbreviated/delayed ringing for each call appearance when per button ring control is enabled. Default is r.
Room
Valid Entry
Telephone location
Guest room number
Usage
Identifies the telephone location. Accepts up to 10 characters.
Identifies the guest room number if this station is one of several to be assigned a guest room and the Display Room Information in Call Display is enabled for the system. Accepts up to five digits.
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Station
Speaker
Indicates whether or not the station is equipped with a speaker.
Set Color
Indicates the set color. Valid entries include the following colors: beige, black, blue, brown, burg (burgundy), gray, green, ivory, orng (orange), red, teak, wal (walnut), white, and yel (yellow).
You can change the list of allowed set colors by using the Valid Set Color fields on the site-data screen.
Voice mail Number
Valid Entries digits (1 to 9) * # ~p (pause) ~w/~ (wait) ~m (mark) ~s (suppress)
Usage
Provides a voice mail retrieval AUX dial button on 4xx, 46xx, and 96xx telephones. When a number is entered in this field, the telephone's fixed voice mail retrieval button acts as an autodial button, dialing the number entered in this field to access voice mail. Accepts up to 24 digits.
Note:
If this field is left blank, the telephone's fixed voice mail retrieval button acts as a "transfer to voice mail" button that only works for Intuity Audix or QSIGintegrated voice mail systems. For Intuity Audix and QSIG-integrated voice mail systems, leave the field blank. For non-QSIG integrated voice mail systems, set the field to an appropriate number.
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BUTTON ASSIGNMENTS
The feature assigned to each button on the station. Feature buttons are assigned by entering the abbreviated feature. For a list of feature buttons, see Telephone Feature Buttons Table in Administering Avaya Aura® Communication Manager.
Note: To use Terminal Translation Initialization (TTI), a call appearance (call-appr) must be assigned to the first button position. TTI needs the button on the first call appearance to get dial tone.
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Data Extension
Use this field to assign an extension to the data module. You can enter up to five digits. This value must conform to the dial plan.
Default Dialing Abbreviated Dialing Dial Code
The list number associated with the abbreviated dialing list. When the user goes off-hook for a data call and presses the Enter button following the DIAL prompt, the system dials the AD number. Available only if the Special Dialing Option is set to default.
1-Step Clearing
Enables or disables call termination at the WCBRI terminal when the user drops from the call.
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Enable Reachability for Station Domain Control
Using the Enable Reachability for Station Domain Control field, you can enable or disable options polling for specific stations.
Note: When the system-parameters features form is set to disable-all, the station form field is overridden or made inaccessible. Enable Reachability for Station Domain Control field has the following options:
Valid Entry y n
Usage To enable the polling feature for an individual SIP station. To disable the polling feature for an individual SIP station.
Table continues...
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Station
Valid Entry system
Usage To take action based on the option selected on the system-parameters features form, type system.
Note:
The default option is set to system. The administration option has effect only if the station is domain-controlled.
Primary Session Manager and secondary Session Manager
These two fields are populated by System Manager and are read only on Communication Manager. You cannot modify these fields from Communication Manager SAT interface. If the call to this SIP station is routed to a SIP trunk that has a clustered signaling group, then SIP Trunk Optimization feature sends the Invite to the station on primary Session Manager. If primary Session Manager is not reachable, then the Invite will be sent to the secondary Session Manager. The Session Manager selection for routing to a SIP station does not apply to a call between two SIP stations. In this case, the Session Manager itself decides the destination Session Manager for the called SIP station.
Route Selection
This field displays the value of the Trunk Selection of the corresponding off-pbx-telephone station mapping entry if there is one. The valid entries are:
· ars · aar · Trunk group number: tg1 to tg2000 · Route pattern number: rp1 to rp2000
SIP URI
Use this field to place a call using the URI. You can enter up to 70 characters. You can use letters, numbers, and special characters.
Following are not supported:
· Unicode characters
· Eurofont characters
· Using same URIs across hunt groups, VDNs, and stations within the same Communication Manager.
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Example support-helpdesk@example.com
Automatic Selection of DID Numbers
Assigned Member -- Name
The name associated with the extension of the user who has an associated Data Extension button and shares the module. This field is available only if the station type is 6408D and data module is set to y.
Assigned Member -- Ext
The extension of the user who has an associated Data Extension button and shares the module. This field is available only if the station type is 6408D and data module is set to y.
Type of 3PCC Enabled
Indicates if the telephone can be controlled by an Avaya third Party Call Control (3PCC) adjunct or a Computer Telephony Integration (CTI) adjunct. The valid entries are none and Avaya.
Automatic Selection of DID Numbers
Enables or disables the Automatic Selection of DID Numbers for Guest Rooms feature. This feature assigns a two- to five-digit extension from a predetermined list of numbers to a hotel room telephone number that is disassociated with the room number.
Country Protocol
The protocol that corresponds to the supported initialization and codesets. The Country Protocol must match any previously-administered endpoint on the same port.
Coverage Module
Indicates whether or not a coverage module is connected to the station.
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Station
Custom Selection of VIP DID Numbers
The selection or removal of a DID number assigned to a room when a guest checks in. Available only if Automatic Selection of DID Numbers is enabled. Related links Automatic Selection of DID Numbers on page 382
Data Option
Valid Entry analog data module none
Usage A second line on the telephone is administered on the I-2 channel. A second line on the telephone is not administered on the I-2 channel. The data option is not administered.
DSN Number
You can enter a 10-digit extension, or leave the field blank.
Feature Module
Indicates whether or not the station is connected to a feature module.
Idle Appearance Preference
Indicates which call appearance is selected when the user lifts the handset and there is an incoming call.
Valid Entry y n
Usage
The user connects to an idle call appearance instead of the ringing call.
The Alerting Appearance Preference is set and the user connects to the ringing call appearance.
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Message Server Name
Message Server Name
Specifies which Message Server is associated with the station. Must contain a user-defined adjunct name that was previously administered. Names must be administered in alphabetical order.
Public (PSTN) Number
You can enter an 11-digit extension, or leave the field blank.
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Chapter 121: Stations With Off-PBX Telephone Integration
Maps an office telephone to a cell phone through the Extension to Cellular feature. The office telephone can be a standard office number or an administration without hardware (AWOH) station. Example command: add off-pbx-telephone station-mapping
Stations With Off-PBX Telephone Integration: page 1
Application
The type of off-PBX application that is associated with the office telephone. More than one application can be assigned to an office telephone.
Valid Entry blank CSP EC500 HEMU OPS PBFMC PVFMC SCCAN SPFMC VEMU VIEMU ONE-X
Usage Default is blank. cell phone with Extension to Cellular provided by the cellular service provider cell phone with Extension to Cellular Home Enterprise Mobility User SIP enabled phone Public Fixed Mobile Convergence Private Fixed Mobile Convergence wireless SIP phone and cell phone Service Provider Fixed-Mobile Convergence Visited Enterprise Mobility User Visited Initial Enterprise Mobility User It is used by Avaya one-X® Client Enablement Services.
You cannot enter any data for the ONE-X application types, but you can see this on the display command if it is active.
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Stations With Off-PBX Telephone Integration: page 1
CC
The country code associated with the extension. Accepts up to three digits. Multiple entries that use the same telephone number must also have the same country code.
Country code changes made to existing stations or XMOBILE entries are applied to all instances of the telephone number.
SAFE (Self-Administered Feature Access Code for EC500) must not be used with an extension that has an administered country code. Origination mapping can occur with or without a country code. Default is blank.
Configuration Set
Valid Entry 1 to 99
blank
Usage
The Configuration Set number that contains the required call treatment options for the Extension to Cellular station.
The SCCAN application requires two different configuration sets selected for each station. The first set is the value for the WLAN followed by a slash. The second is the value for the cellular network.
Enterprise Mobility User (EMU) and ONE-X application type display a blank entry.. The default setting is blank.
Dial Prefix
The dial prefix the system prepends to the off-switch telephone number before dialing the offswitch telephone. The system deletes the dial prefix when a user enters their cell phone number using the Self Administration Feature (SAFE) access code. The routing tables must be set properly so that the dial prefix 1 is not necessary for correct routing. Communication Manager inserts a plus (+) digit before initiating a call to a SIP endpoint or a cell phone.
· asterisk (*) or hash (#) must be in the first digit position
· 1 must be used if the telephone number is long-distance
· 011 must be used if the telephone number is international
Dual Mode
The Dual Mode field provides functionality for dual mode wireless phones that operate both in the public cellular network using GSM or CDMA and in a wireless network using SIP. The Dual Mode field indicates which application is part of a dual-mode pair. The dual Mode pair determines the pair of channels that the wireless phone is using, one channel over GSM/CDMA and the other
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Stations With Off-PBX Telephone Integration
channel over SIP. The dual Mode pairing is required to transfer a call to the correct channel when the call is moving between GSM/CDMA and SIP networks.
Valid Entry blank DM1
DM2
DMX
Usage
The application is not part of a dual-mode pair.
The application belongs to a dual-mode pair. Applications of the same pair have the same value.
The application belongs to a dual-mode pair. Applications of the same pair have the same value.
Supported only for Private Fixed Mobile Convergence application for pairing with one-X application.
Note:
dm1 or dm2 can be assigned to any two of the Public Fixed Mobile Convergence (PBFMC), Private Fixed Mobile Convergence (PVFMC), and Service Provider Fixed-Mobile Convergence (SPFMC) applications. However, one of the applications must be PVFMC. The other application can be PBFMC or SPFMC.
Phone Number
The Phone Number field accepts: · The telephone number of the off-switch telephone. · The telephone number of an on-PBX extension for the one-X® application type.
The Phone Number field can be blank for: · the first EC500, CSP, or PBFMC telephones you administer · EC500, CSP, and PBFMC telephones that support self-administered Feature Access Code for EC500 (SAFE)
Station Extension
The number of the office telephone that is mapped to the cell telephone. Accepts up to sixteen digits. This number must be an administered extension. Default is blank.
Trunk Selection
The Trunk Selection field indicates the trunk group selection method used for the outgoing call to Session Manager only if the application field is set to ops. If the application field is set to any other value, for example EC500, then the selected trunk will not be Session Manager. The selected trunk will be some other destination, for example, to a cell phone in the PSTN. In the half-call
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Stations With Off-PBX Telephone Integration: page 2
model, indicates a different trunk group than the trunk group that eventually connects the call to the telephone network. The valid options are:
· AAR · ARS · TG1 to TG2000 · RP1 to RP2000
Stations With Off-PBX Telephone Integration: page 2
Bridged Calls
Determines if bridged call appearances extend to the Extension to Cellular cell telephone.
Note: For an Avaya SIP AST phone, this field must be set to none.
Valid Entry termination
origination both none
Usage
Users can use their Extension to Cellular cell phone to only receive calls from the associated office telephone. Users cannot use the cell phone to originate calls from the associated office telephone. Calls originating from the cell phone independent of the office phone are independent of Extension to Cellular and behave exactly as before enabling Extension to Cellular.
Note:
The on-PBX extensions that the ONE-X application type enables are in termination mapping mode only.
Users can originate Extension to Cellular cell phone calls only from the associated office telephone. Users cannot use the cell phone to receive calls from the associated office telephone.
Users can originate Extension to Cellular cell phone calls from the associated office telephone, as well as receive calls from the associated office telephone.
Users cannot originate or receive calls from the office telephone with the cell phone. This is the default value.
Call Limit
Use this field to set the maximum number of calls that can be made simultaneously by using an application, such as Extension to Cellular (EC500) or Off-Premise Station (OPS), on a station. You can enter a value from 1 to 10. For example, if you set the Call Limit field for an EC500
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Stations With Off-PBX Telephone Integration
application to 2 and the Call Limit field for an OPS application to 4, the station can have up to two EC500 calls and up to four OPS calls.
Different scenarios are as follows:
· For SIP station type, the call limit is not administered using the OPS form. Instead, the call limit is automatically counted by the call appearance and bridge button administered on the Station screen. In this scenario, the Station screen is used for SIP station only.
· For H.323 station type that has an associate SIP configuration, the call limit for OPS station is administered using the OPS form, which is independent of the button admin on the Station screen. The Station screen is used for H.323 configuration only. For SIP configuration, administration is done using the OPS form.
· For EC500, the call limit is administered using the OPS form for both SIP and H.323 stations.
Calls Allowed
Identifies the call filter type for an Extension to Cellular station. Determines the type of calls to the office telephone that a user can receive on an Extension to Cellular cell telephone.
Valid Entry all
internal external none
Usage The cell telephone receives both internal and external calls.
Default is all. The cell telephone receives only internal calls. The cell telephone receives only external calls. The cell telephone does not receive any calls made to the associated office telephone.
Location
Valid entry 1 to 2000
blank
Usage
(Depending on your server configuration, see Avaya Aura®Communication Manager System Capacities Table.) The location value for each administered OPS, PBFMC, SPFMC or PVFMC application. Typically, set this to the same location number being used by the desk set. For more information on location, see Avaya Aura® Communication Manager Feature Description.
Trunk location is used for the outgoing calls from Off-PBX endpoints and location of station is used for incoming calls to Off-PBX endpoints. Blank is the default.
Related links Application on page 790 Loc Number on page 566
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Stations With Off-PBX Telephone Integration: page 3
Mapping Mode
The mode of operation for the Extension to Cellular cell phone. These modes control the degree of integration between the cell phone and the office telephone. The modes are valid for Extension to Cellular calls only. For each office telephone, only one cell phone can be assigned as the origination mode. A cell phone cannot be assigned as either the origination or both mode more than once.
Valid Entry termination
origination both none
Usage
Users can use their Extension to Cellular cell phone to only receive calls from the associated office telephone. Users cannot use the cell phone to originate calls from the associated office telephone. Calls originating from the cell phone independent of the office phone are independent of Extension to Cellular and behave exactly as before enabling Extension to Cellular.
Note:
The on-PBX extensions that the ONE-X application type enables are in termination mapping mode only.
Users can originate Extension to Cellular cell phone calls only from the associated office telephone. Users cannot use the cell phone to receive calls from the associated office telephone.
Users can originate Extension to Cellular cell phone calls from the associated office telephone, as well as receive calls from the associated office telephone.
Users cannot originate or receive calls from the office telephone with the cell phone. This is the default value.
Stations With Off-PBX Telephone Integration: page 3
Share Level
Indicates the sharing level. Enter the value from 1-9.
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Chapter 122: Survivable ARS Analysis Table
Communication Manager compares dialed numbers with the dialed strings in this table and determines the route pattern for the number. Example command: change survivable-ars-analysis
Call Type
Valid Entry emer npa hnpa intl iop locl natl op svc
Usage The call type used for the dialed string.
Deny
Indicates whether or not the dialed string should be blocked. The system denies a dialed string that does not match an entered pattern.
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Dialed String
Dialed String
Communication Manager matches the dialed numbers with the entry in the Dialed String field that most closely matches the dialed number. You can enter up to 18 digits that the call-processing server analyzes. You can also enter the wildcard characters, x and X.
Total Length
Valid Entry 0 to 28
Usage
The minimum number of digits required to validate the route. The minimum value when the dial string is populated is the length of the dialed string entry with a maximum value up to 28. Default is blank.
Trunk Grp No
The trunk group number that specifies the destination route for the dial plan analysis of this dialed string.
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Chapter 123: Survivable Processor
Adds information specific to a Survivable Remote Server (Local Survivable Processor) or to connect certain adjuncts to a Survivable Remote or Survivable Core server (Enterprise Survivable Server). Before administering this screen, you must first assign node names for each Survivable Remote and Survivable Core media server on the IP Node Names screen. While this screen is administered on the active main server, the information entered applies only to Survivable Remote and Survivable Core servers and does not apply to main servers. When translations are copied to a Survivable Remote or Survivable Core server, the Survivable Remote Server/Survivable Core Server replaces like translations for the main server with the overrides administered on the Survivable Processor screens. That is, use the Survivable Processor screen to administer overrides against adjunct links that have already been administered for the main servers. For more information about survivable core server, see Avaya Aura® Communication Manager Survivable Options. Example command: add survivable-processor n, where n is the assigned node name.
Survivable processor: page 1
Cluster ID
Valid Entry 1 to 999
Usage
The Cluster ID for the survivable core server. The Cluster ID corresponds to the Module ID as found on the Server Role web page of the Survivable Core Server or Survivable Remote Server.
Note:
The statuslicense -v shell command on the survivable core server or the survivable remote server displays the Module ID.
Community
A community is a virtual group consisting of a survivable core server and one or more port networks. Assigning a survivable core server to a community associates the survivable core server with each IPSI in each port network for that community. Each IPSI is assigned to
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Survivable processor: page 1
communities system-wide. The association affects how the survivable core server is prioritized for the IPSI in that community, if the survivable core server is administered with a Local Preferred or Local Only preference. The Community number for an S8400 survivable core server must be set to 2 or greater and must be unique. Related links System Parameters Port Networks on page 862
Community Size
Valid Entry Sngl_PN all
Usage For a survivable core server, the value must be Sngl_PN. Default is all.
Enable PE for H.248 Gateways
Enables or disables using the PE interface of the survivable core server for gateways.
Enable PE for H.323 Endpoints
Enables or disables using the PE interface of the survivable core server for H.323 devices such as telephones.
IP Address
The IP address that corresponds to the node name.
There are three IP addresses, one for each node name if the survivable processor is a duplicated survivable core.
Related links Name on page 475
Local Only
Enables or disables the survivable core server accepting a request for service from an IPSI. Can be enabled only if the IPSI is located in the same community as the survivable core server. Automatically enabled for survivable core servers and cannot be changed. Otherwise, this field defaults to disabled.
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Survivable Processor
Related links System Preferred on page 801 Local Preferred on page 800
Local Preferred
Valid Entry y
n
Usage
The survivable core server accepts a request for service from IPSIs co-located in the same geographical region, WAN segment, LAN segment, district, or business unit.
Default when the community size is set to Sngl_PN for an S8400 survivable core server and cannot be changed.
Related links Community Size on page 799 System Preferred on page 801 Local Only on page 799
Node Name
The previously-administered name used to identify the server. If the survivable processor is duplicated, there are three node names, one each for the duplicated server pair and one for the server that is active at a given point.
Priority Score
Valid Entry 1 to 100
Usage The priority score for this survivable core server. Default is 1.
Processor Ethernet Network
Valid Entry 1 to 2000
Usage
The network region in which the PE interface of the Survivable Remote or Survivable Core server resides.
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Survivable Processor: page 3 (IP Services page)
Server A -- Server ID
Valid Entry 1 to 256
Usage
Server A ID corresponds to the Server ID configured on the survivable core server. The administration on the main server and the configuration on the survivable core server must match for the survivable core server to register to the main server.
Server B -- Server ID
Valid Entry 1 to 256
Usage For duplicated survivable core servers, displays the node name of Server B.
System Preferred
Valid Entry y
n
Usage
Allows one survivable core server to replace the main server to keep as much of the system network intact as possible. Disables Local Preferred and Local Only and cannot be changed. This is the default except for an S8400 survivable core server.
The default for an S8400 survivable core server and cannot be changed.
Related links Local Only on page 799 Local Preferred on page 800
Type
Valid Entry lsp simplex_ess duplex_ess
Usage The survivable processor type.
Survivable Processor: page 3 (IP Services page)
Used when an AESVCS or a CDR connects to the Survivable Remote or Survivable Core server.
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Survivable Processor
Enabled
Specifies how data for each specified service type is transferred to the survivable processor.
Valid Entry i(nherit)
n(o) o(verwrite)
Usage
This link is to be inherited by the Survivable Remote or Survivable Core server. The survivable processor inherits this service type just as it is administered on the main server. Used in the following situations:
· The main server connects to the adjuncts using a CLAN and you want the Survivable Core Server to use the same connectivity
· The main server connects to the adjuncts using the main server's PE interface and you want the Survivable Remote or Survivable Core server to connect to the adjunct using its PE interface
This IP services link is disabled on the Survivable Remote or Survivable Core server. This is the default.
Overwrites the processor channel information sent in the file sync from the main server. With the entry o(overwrite), the administered CDR or AE Services attributes can be modified uniquely for each individual Survivable Remote or Survivable Core server.
Local Node
Displays the previously-administered node name. Related links
Name on page 475
Local Port
The originating port number. For client applications such as Call Detail Recording (CDR), this field defaults to 0.
Priority with respect to Media Servers
Use this field to assign the priorities to the media servers. The priority of main server is always 1, so you must enter a priority value from 2 to 9999.
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Survivable Processor: page 4 (IP Services -- Session Layer Timers page)
Remote Node
The name at the far end of the link for the CDR. Does not apply for AESVCS.
Remote Port
Valid Entry 5000 to 65500
Usage
The port number of the destination for CDR or AESVCS. This number must match the port administered on the CDR or AESVCS server.
Service Type
Valid Entry CDR1 CDR2 AESVCS
Usage The service provided.
Related links Service Type on page 500
Store to dsk
Enables or disables the storage of the CDR data on the local hard drive of the Survivable Remote or Survivable Core server. Pertains only to service types set to CDR1 or CDR2.
Survivable Processor: page 4 (IP Services -- Session Layer Timers page)
Available only if CDR1 or CDR2 is administered, and if processor channel information is administered to be overwritten.
Related links Service Type on page 803 Enabled on page 802
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Survivable Processor
Connectivity Time
Valid Entry 1 to 255
Usage
The amount of time that the link can be idle before Communication Manager sends a connectivity message to ensure the link is still up. Default is 60.
Packet Resp Timer
Valid Entry 1 to 255
Usage
The number of seconds to wait from the time a packet is sent until a response (acknowledgement) is received from the far-end, before trying to resend the packet. Default is 30.
Reliable Protocol
Enables or disables using a reliable protocol over this link. A reliable protocol should be used if the adjunct on the far end of the link supports it.
Service Type
Displays the previously-administered service type. Related links
Service Type on page 803
Session Connect Message Cntr
Valid Entry 1 to 5
Usage
The number of times Communication Manager tries to establish a connection with the far-end adjunct.
SPDU Cntr
Valid Entry 1 to 5
Usage
The number of times Communication Manager transmits a unit of protocol data before generating an error.
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Chapter 124: System Capacity
Provides a status of administered capacity information and a snapshot status of system resources. For more information, see Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and Servers. Detailed system capacity information can be found in Avaya Aura® Communication Manager System Capacities Table.
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Chapter 125: System Configuration
The System Configuration screen shows all the boards on the system that are available for connecting telephones. Used to view the board number, board type, circuit-pack type, and status of each board's ports.
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Chapter 126: System Parameters Call Coverage/Call Forwarding
Sets the system-wide parameters for call coverage and call forwarding. Example command: change system-parameters coverage-forwarding
System parameters call coverage/ call forwarding: page 1
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System Parameters Call Coverage/Call Forwarding
CALL COVERAGE / FORWARDING PARAMETERS
Coverage - Caller Response Interval (seconds)
Valid Entry 0 to 10
Usage
The time in seconds an internal caller has before a call redirects to the called party's first coverage point.
COR/FRL check for Covered and Forwarded Calls
Valid Entry y
n
Usage Communication Manager checks the following:
· COR of the calling and forwarded-to parties for local forwarding.
· COR and FRL of the calling party and outgoing trunk for remote forwarding and remote coverage calls.
Communication Manager does not check the following:
· COR of the calling and forwarded-to parties for local forwarding.
· COR and FRL of the calling party and outgoing trunk for remote forwarding and remote coverage calls. This is the default value.
Diverted Party Identification
The Diverted Party Identification field is available only if the QSIG/SIP Diverted Calls Follow Diverted to Party's Coverage Path field is enabled. The default setting is last-fwd.
The field settings do not work if the incoming call is over a non-QSIG or non-SIP trunk or the call is routed through a route-to vector step.
Valid entry principal
last-fwd
Usage
To play voice mail greeting of the principal and save the voice message at the voicemail box of the principal.
To play voice mail greeting of the last-forwarded party and save the voice message at the voicemail box of the last-forwarded party. The default configuration.
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System parameters call coverage/ call forwarding: page 1
Local Cvg Subsequent Redirection/CFWD No Ans Interval (rings)
Valid Entry 1 to 99
Usage
Specifies:
· The number of rings applied at a local coverage point before a call redirects to the next coverage point
· The number of rings applied at the principal before a call forwards when Call Forwarding Busy/Don't Answer is activated
Note:
When ringing local destinations, such as an office environment, a short interval often is appropriate because the intended party either is near the telephone or not present. However, if the call is left at an off-net destination for only a short interval, the call can be redirected to the next destination before the intended party has any real chance of answering the call.
Location for Covered and Forwarded Calls
Determines the location number used for coverage and forwarding.
Valid Entry called
caller
Usage · If the called party is registered or in-service, coverage and forwarding use the
called party's physical phone's location number.
· If the called party is AWOH (x-port) or unregistered, coverage and forwarding use a location based on the type of set.
- IP set: location 1
- DCP set: location 0 (all)
· When the forwarding or coverage destination is to UDP instead of to an external destination starting with the ARS FAC, routing is always based on the caller's physical phone's location regardless of how this field is administered.
This is the default. Coverage and forwarding use the caller's physical phone's location number.
Related links ARS FAC on page 564
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System Parameters Call Coverage/Call Forwarding
Off-Net Cvg Subsequent Redirection/CFWD No Ans Interval (rings)
Valid Entry 1 to 99
Usage
Specifies:
· The number of rings applied at an off-net coverage point before a call is redirected to the next coverage point
· The number of rings applied at an off-net forwarded-to destination before the call is redirected to coverage.
Note:
When ringing local destinations, such as an office environment, a short interval often is appropriate because the intended party either is near the telephone or not present. However, if the call is left at an off-net destination for only a short interval, the call can be redirected to the next destination before the intended party has any real chance of answering the call.
PGN/TN/COR for Covered and Forwarded Calls
Valid Entry called
caller
Usage
Communication Manager checks for permissions (tenant number, partition group number, and COR) between the actual called party (forwarding party) and forwarded-to party. When the check is successful, the call is routed to the forwardedto party. Permission check for COR is only done if the COR/FRL check for Covered and Forwarded Calls field is set to y.
Communication Manager checks for permissions (tenant number, partition group number, and COR) between the calling party and forwarded-to party. When the check is successful, the call is routed to the forwarded-to party. Permission check for COR is only done if the COR/FRL check for Covered and Forwarded Calls field is set to y. This is the default value.
QSIG/SIP Diverted Calls Follow Diverted to Party's Coverage Path
When a party with a defined coverage path does not answer the call, Communication Manager determines whether to continuously ring the call or redirect the call to a station or a messaging system. The calls follow the coverage path of the diverted-to party only if the calls route over QSIG and SIP trunks.
Valid entry y
n
Usage
Communication Manager forwards the call to the coverage path of the last redirected party. The Diverted Party Identification field displays the default value as last-fwd.
The default configuration. The call continuously rings at the extension of the redirected party even if that extension has a coverage path. The call does not reach the messaging system of the last party in the coverage path.
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System parameters call coverage/ call forwarding: page 1
Threshold for Blocking Off-Net Redirection of Incoming Trunk Calls
Valid Entry 1 to 7
n (all)
Usage
If the number of incoming trunk calls routes within the Call Forward timer, the block commences. Applies for those occasions when an incoming call to a station redirects off-net. At that time, the Call Forward timer activates to block any further incoming calls to that station from being redirected off-net until the timer expires.
Call processing never activates the Call Forward timer. Therefore, any number of calls to a principal can be redirected off-net.
COVERAGE
Criteria for Logged Off/PSA/TTI Stations
Enables or disables the call coverage criteria for logged-off IP/PSA/TTI stations. By default, the value is n.
Default treatment for logged off phones is a busy tone. If you set the Criteria for Logged Off/PSA/TTI Stations field to y, calls follow the coverage treatment as configured on the station screen. The call is redirected to coverage after the number of rings exceed the number specified in the Number of Rings field for Criteria for Logged Off/PSA/TTI Stations.
Related links Active on page 175
External Coverage Treatment for Transferred Incoming Trunk Calls
Enables or disables external coverage treatment for incoming trunk calls that redirect to coverage.
Immediate Redirection on Receipt of PROGRESS Inband Information
This field determines if a call is immediately redirected to coverage when an ISDN PROGRESS message is received. This field pertains only to CCRON and QSIG VALU coverage calls redirected over end-to-end ISDN facilities.
However, the message might indicate that the cell telephone is not available to receive calls and should be redirected.
The system displays this field only if:
· The Coverage of Calls Redirected Off-Net field shows y. · The Value-Added Avaya (VALU) feature is enabled for the system.
Valid entry y
Usage
Immediately redirect an off-net coverage/forwarded call to the next coverage point. Users in European countries following the ETSI standard and redirecting to GSM cellular telephones should enable the field.
Table continues...
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System Parameters Call Coverage/Call Forwarding
Valid entry n
Usage
Does not immediately redirect an off-net coverage/forwarded call to the next coverage point. Users in the United States should disable this field because PROGRESS messages with the Progress Indicator field set to inband information are sent for a variety of reasons that are not associated with unavailable cellular telephones.
Related links Cvg Of Calls Redirected Off-net on page 831 Value Added (VALU) on page 849
Keep Held SBA at Coverage Point
Determines how a covering user who has placed an answered coverage call on hold is treated if the original principal bridges onto the call.
Valid Entry y
n
Usage
Keeps the coverage party on the call. The coverage party remains on hold, but might enter the call along with the principal and the calling party.
Drops the coverage party from the call.
Maintain SBA At Principal
Using this, a user can maintain a simulated bridged appearance (SBA) when a call redirects to coverage.
Valid Entry y
n
Usage
Maintains a simulated bridged appearance on the principal's telephone when a call redirects to coverage. DCS with rerouting will not be attempted after coverage.
No SBA is maintained on the principal's telephone. DCS with rerouting is attempted, and if successful, the principal loses the bridged appearance and the ability to bridge onto the coverage call.
QSIG VALU Coverage Overrides QSIG Diversion with Rerouting
Specifies whether or not, with both QSIG Diversion with Rerouting and QSIG VALU turned on, Coverage After Forwarding for the station works for calls that go to remote coverage.
Available only if Basic Supplementary Services and Supplementary Services with Rerouting are both enabled for the system.
Valid Entry y
n
Usage
QSIG VALU call coverage takes precedence. If Coverage After Forwarding is enabled for a station, the call can receive coverage after rerouting.
With QSIG Diversion with Rerouting turned on, the local system passes control of a forwarded call to the remote QSIG server on which the forwarding destination resides. The forwarded call cannot return to coverage for the user who originally received the call.
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System parameters call coverage/ call forwarding: page 1
Related links Basic Supplementary Services on page 848 Supplementary Services with Rerouting on page 849
Station Hunt Before Coverage
Determines whether or not a call to a busy station performs station hunting before going to coverage.
FORWARDING
Call Forward Override
Specifies how to treat a call from a forwarded-to party to the forwarded-from party.
Valid Entry y
n
Usage
Overrides the Call Forwarding feature by allowing a forwarded-to station to complete a call to the forwarded-from station.
Directs the system to forward calls to the forward-to station even when the calls are from the forwarded-to station.
Coverage After Forwarding
Determines whether or not an unanswered forwarded call is provided coverage treatment.
Valid Entry y n
Usage
Coverage treatment is provided to unanswered forwarded calls.
No coverage treatment is provided to unanswered forwarded calls. The call remains at the forwarded-to destination.
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System Parameters Call Coverage/Call Forwarding
System parameters call coverage/ call forwarding: page 2
COVERAGE OF CALLS REDIRECTED OFF-NET (CCRON)
Activate Answer Detection (Preserves SBA) On Final CCRON Cvg Point
Determines whether or not a simulated bridge appearance (SBA) is maintained on the principal when a call is directed to a final off-net coverage point. This field has no consequence when the off-net call is carried end-to-end by ISDN facilities; the SBA is maintained and there is no cutthrough delay.
Available only if Coverage of Calls Redirected Off-Net Enabled is enabled.
Valid Entry y
n
Usage
Maintains a simulated bridged appearance on the principal when redirecting to a final off-net coverage point.
Drops the SBA on the principal's telephone when the call redirects off-net at the last coverage point, eliminating the cut-through delay inherent in CCRON calls, but sacrificing the principal's ability to answer the call.
Related links Coverage Of Calls Redirected Off-Net Enabled on page 815
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System parameters call coverage/ call forwarding: page 2
Coverage Of Calls Redirected Off-Net Enabled
Controls the Coverage of Calls Redirected Off-Net (CCRON) feature. Disables this feature if the demand on the call classifier port resources degrades other services provided by Communication Manager.
Available only if Coverage of Calls Redirected Off-Net is enabled for the system.
Valid Entry y
n
Usage
Communication Manager monitors off-net coverage/forwarded calls and provides further coverage treatment for unanswered calls.
Communication Manager does not monitor off-net coverage/ forwarded calls. No further coverage treatment is provided if the calls are unanswered.
Related links Cvg Of Calls Redirected Off-net on page 831
Disable call classifier for CCRON over ISDN trunks
Enables or disables the use of a call classifier on a CCRON call over ISDN facilities. When a CCRON call routes offnet over ISDN end-to-end facilities, no call classifier is attached to the call. If, subsequently during the call, an ISDN PROGRESS or ALERT message is received that indicates that interworking has occurred, a call classifier is normally attached to the call and has precedence over ISDN trunk signalling. This field can direct Communication Manager to dispense with the call classifier on interworked calls and rely on the ISDN trunk signalling messages.
Valid Entry y n
Usage Disables the call classifier for CCRON calls over interworked trunk facilities. Enables the call classifier for CCRON calls over interworked trunk facilities.
Disable call classifier for CCRON over SIP trunks
Valid Entry y
n
Usage
Disables the call classifier for CCRON calls over interworked trunk facilities. Directs Communication Manager to dispense with the call classifier on interworked calls and rely on the SIP trunk signalling messages.
Enables the call classifier for CCRON calls over interworked trunk facilities.
Ignore Network Answer Supervision
Administers whether or not a call classifier for network answer supervision is used to determine when a call is answered. CCRON might use a call classifier port to determine whether an off-net coverage or forwarded call has been answered, discarding other information that might indicate an answered state.
Available only if Coverage of Calls Redirected Off-Net Enabled is enabled.
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System Parameters Call Coverage/Call Forwarding
Valid Entry y
n
Usage
Ignore network answer supervision and rely on the call classifier to determine when a call is answered. Provides accurate answer supervision for tandem calls redirected to the public network.
Treat network answer supervision as a true answer. Preserves network answer supervision information.
Related links Coverage Of Calls Redirected Off-Net Enabled on page 815
CHAINED CALL FORWARDING
Maximum Number of Call Forwarding Hops
Available only if Chained Call Forwarding is enabled.
Valid Entry 3 to 10
Usage The number of hops allowed in the forwarding chain.
Related links Chained Call Forwarding on page 351
Station Coverage Path For Coverage After Forwarding
This field specifies what coverage path the call must follow. The system displays the Station Coverage Path For Coverage After Forwarding field only when you enable the Chained Call Forwarding and the Coverage After Forwarding fields.
Valid entry principal last-fwd
Usage
The call follows the coverage path of the principal.
The call follows the coverage path of the lastforward-to party.
Related links Chained Call Forwarding on page 351 Coverage After Forwarding on page 813
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Chapter 127: System Parameters Country Options
This screen implements parameters associated with certain international, including North American, call characteristics. This screen cannot be changed. Go to the Avaya Support website at http:// support.avaya.com for current documentation, product notices, knowledge articles on how to modify any of the values here. This table shows the country codes that are used in Communication Manager. The Country Code is used by various fields and screens throughout the system. Example command: change system-parameters country-options
Country options table
Code 1
Country
United States, Canada, Korea, India
2
Australia, New Zealand
3
Japan
4
Italy
Ringing Signal Voltage, Frequency, and Cadence
300v peak to peak, < 200v peak to ground; < 70 Hz; < 5s on > 1s off
Korea: 20 Hz, 75 to 85 Volts (AC), Cadence: 1 sec on, 2 sec off
75 +/- 20 VRMS superimposed on 48 V dc at 14.5 to 55 Hz with cadence 400ms on, 200ms off, 400ms on, 2000ms off
New Zealand: Ringing voltage at the customer's premises not less than 38 V rms (25Hz) on top of 50V d.c; 20 Hz; 400ms on, 200ms off, 400ms on, 2000ms off
75 VRMS(75-10VRMS <= x <= 75+8VRMS), 15-20 Hz and cadence of 1second on and 2 seconds off is required
20 to 50 Hz, 26 to 80 Volts rms superimposed on 48 V dc, 1 sec on, 4 sec off
ETSI countries: 30 Volts rms, superimposed on a DC voltage of 50 Volts, 25 or 50 hz, cadence of 1 sec on, 5 sec off
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System Parameters Country Options
Code 5
6 7 8
9 10
11
Country Netherlands
Singapore Mexico Belgium, Luxembourg, Korea
Saudi Arabia United Kingdom
Spain
Ringing Signal Voltage, Frequency, and Cadence
25 Hz, 35 to 90 Volts rms superimposed on 66 V dc, 1 sec on, 4 sec off. Note that 50 Hz should be used, and another cadence can be 0.4 sec on, 0.2 sec off, 0.4 sec on, 4 sec off
ETSI countries: 30 Volts rms, superimposed on a DC voltage of 50 Volts, 25 or 50 hz, cadence of 1 sec on, 5 sec off
75V at 24Hz with a cadence of 0.4 seconds on, 0.2 seconds off, 0.4 seconds on and 2.0 seconds off.
25 Hz, 70 +/- 20 Vrms superimposed on 48Vdc Cadence 1 sec on, 4 sec off, flashhook is 100 ms
25 Hz, 25 to 75 Volts rms superimposed on 48 V dc, 1 sec on, 3 sec off
Korea: 20 Hz, 75 to 85 Volts (AC), Cadence: 1 sec on, 2 sec off
ETSI countries: 30 Volts rms, superimposed on a DC voltage of 50 Volts, 25 or 50 hz, cadence of 1 sec on, 5 sec off
U.K.: 15 to 26.25 Hz, 25 to 100 Volts rms superimposed on 48 V dc, 0.35 on, 0.22 off then start in at any point in: 0.4 sec on, 0.2 sec off, 0.4 sec on, 2 sec off. Note 1: 48v DC can be present during the whole cadence or can be confined to silent periods. Note 2: Some exchanges provide a facility known as immediate ring; in this case an initial burst of ringing 20 msec to 1 sec in length immediately precedes switching to any point in the normal ringing cycle.
Ireland: 25 Hz, 30 to 90 Volts rms superimposed on 50 V dc, 0.4 sec on, 0.2 sec off, 0.4 sec on, 2 sec off another possible cadence is 0.375 sec on, 0.250 sec off, 0.375 sec on, 2 sec off.
ETSI countries: 30 Volts rms, superimposed on a DC voltage of 50 Volts, 25 or 50 hz, cadence of 1 sec on, 5 sec off
20 to 30 Hz, 35 to 75 Volts rms superimposed on 48 V dc, 1 to 1.5 sec on, 3 sec off
ETSI countries: 30 Volts rms, superimposed on a DC voltage of 50 Volts, 25 or 50 hz, cadence of 1 sec on, 5 sec off
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Code 12
13
14 15
16 17 18 19
20 21 22 23
Country options table
Country France
Germany
Czech Republic, Slovakia Russia (CIS)
Argentina Greece China Hong Kong
Thailand Macedonia Poland Brazil
Ringing Signal Voltage, Frequency, and Cadence 50 Hz, 28 to 90 Volts rms superimposed on 0.45 to 54 V dc, 1.5 sec on, 3.5 sec off ETSI countries: 30 Volts rms, superimposed on a DC voltage of 50 Volts, 25 or 50 hz, cadence of 1 sec on, 5 sec off Germany: 25 Hz, 32 to 75 Volts rms superimposed on 0 to 85 V dc, 1 sec on, 4 sec off Austria: 40 to 55 Hz, 25 to 60 Volts rms superimposed on 20 to 60 V dc, 1 sec on, 5 sec off +/- 20% ETSI countries: 30 Volts rms, superimposed on a DC voltage of 50 Volts, 25 or 50 hz, cadence of 1 sec on, 5 sec off
25+\-2 Hz, 95+\-5 Volts eff, local call cadence: first ring 0.3-4.5 sec then 1 second on 4 seconds Off toll automatic cadence 1 sec On 2 sec Off toll operator: manual sending 25Hz; 75 Vrms superimposed on 48 Vdc; 1s on 4s off
25Hz +/- 3Hz; 75 +/- 15 Vrms; Harmonic Distortion <= 10%; 1 sec ON, 4 secs OFF 75 +/- 20 VRMS superimposed on -40 to -48 V dc at 25 Hz +/- 10% with cadence 0.4 s on, 0.2 s off, 0.4 s on, 3.0 s off
25Hz +/-2.5Hz; minimum of 40 Vrms; 1s on, 4s off for equipment supporting up to six trunks only otherwise 25Hz +/-2.5Hz; minimum of 70+/-15 Vrms at a continuous emitting condition under no load, overlapping a DC level.
Table continues...
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System Parameters Country Options
Code 24
Country Nordic
25
South Africa
Ringing Signal Voltage, Frequency, and Cadence
Finland: 25 Hz, 35 to 75 Volts rms superimposed on 44 to 58 V dc, 1 sec on, 4 sec off
25 Hz, 40 to 120 Volts rms superimposed on 44 to 56 V dc, 0,75 on, 7,5 off +/- 20 %
25 Hz, 28 to 90 Volts rms superimposed on 24 to 60 V dc, 1 sec on, 4 sec off
25 and 50 Hz, 30 to 90 Volts rms superimposed on 33 to 60 V dc, 1 sec off, 5 sec off
ETSI countries: 30 Volts rms, superimposed on a DC voltage of 50 Volts, 25 or 50 hz, cadence of 1 sec on, 5 sec off
Dial Tone Validation Timer (sec)
Available only when Tone Detection Mode is 4 or 5. Valid with TN420C or later Tone Detector circuit pack.
Valid Entry 0 to 6375
Usage
Displays number of milliseconds in increments of 25 that the dial tone validation routine uses to sample transmissions.
Related links Tone Detection Mode on page 822
Disconnect on No Answer by Call Type
Enables or disables the system from disconnecting calls that are not answered. Drops outgoing trunk calls, except DCS and AAR, that users leave unanswered too long.
Directory Search Sort Order
Available only for the Cyrillic or Ukrainian display character set.
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Display Character Set
Valid Entry Cyrillic
Roman
Usage
Cyrillic Collation is used for integrated directory name search and result sorting. This is the default value.
Eurofont Latin Collation is used for directory name search and result sorting. The letters to be searched in the specified order for dial pad button presses are defined in the row for each key.
Related links Display Character Set on page 821
Display Character Set
Valid Entry Cyrillic Katakana Roman Ukranian
Usage
The character set used for all non-native name values without an ASCII-only restriction.
Note:
Cyrillic, Roman, and Ukrainian map to the Eurofont character set. For Katakana, the Optrex font is used. If a Communication Manager server uses non-English in any name field, characters on a BRI station are displayed incorrectly.
Warning:
Changing the value in this field might cause some telephones to perform improperly, and can cause non-ASCII data in non-native names to display incorrectly on telephones. To correct this, non-native names of previously administered stations must be removed and readministered using non-ASCII characters. This includes any display messages that have been administered.
Enable Busy Tone Disconnect for Analog Loop-start Trunks
Enables or disables Busy Tone Disconnect. When enabled, Communication Manager recognizes a busy tone from the local telephone company central office as a disconnect signal.
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System Parameters Country Options
Howler After Busy
Enables or disables howler tone when users leave their analog telephone off-hook too long.
Set Layer 1 timer T1 to 30 seconds
Specifies whether or not the Layer 1 timer is set to 30 seconds.
TONE DETECTION PARAMETERS
Interdigit Pause
Specifies the maximum length of the inter-digit pause. Breaks lasting less than this range will be bridged or ignored. (Valid with TN420C or later Tone Detector circuit pack.)
Valid Entry short long
Usage 530ms 2040ms
Tone Detection Mode
Use this field to specify the type of tone-detection algorithm that Communication Manager must use for a TN420B or later tone-detection circuit pack.
Valid entry 1 2 3 4
5
6
Usage
The precise Italian tone-detection algorithm.
The precise Australian tone-detection algorithm.
The precise UK tone-detection algorithm.
The imprecise normal broadband-filter algorithm. This option is valid only for a TN420C or later tone-detection circuit pack.
The imprecise wide broadband-filter algorithm. This option is valid only for a TN420C or later tone-detection circuit pack.
The precise USA tone-detection algorithm.
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Chapter 128: System Parameters Customer Options
Shows which optional features are enabled for the system, as determined by the installed license file. All fields on this screen are display only. For more information about disabling or enabling one of the features, go to Avaya Support website at http://support.avaya.com/. Example command: display system-parameters customer-options
System parameters customer options: page 1
G3 Version
Identifies the version of Avaya Communication Manager being used.
Location
Indicates the location of this Avaya server or switch.
Valid Entry 1 2
Usage
Canada or the United States
Any other location. Allows the use of International Consolidation circuit packs and telephones.
Maximum Off-PBX Telephones - EC500
Stations that are administered for any Extension to Cellular (EC500/ CSP) application count against this limit.
The "license max" value is defined as follows:
· On legacy systems, the upper limit is 1/2 of the maximum number of administrable stations. Legacy platforms do not support SIP trunks.
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System Parameters Customer Options
· On Linux systems, the upper limit is the maximum number of administrable stations.
Maximum Off-PBX Telephones - OPS
Stations that are administered for any SIP Extension to Cellular/OPS application count against this limit. The "license max" value is defined as follows:
· On legacy systems, the upper limit is 1/2 of the maximum number of administrable stations. Note that legacy platforms do not support SIP trunks.
· On Linux systems, the maximum number of administrable stations.
Maximum Off-PBX Telephones -- PBFMC
Number of stations administered for Public Fixed-Mobile Convergence. Each station is allowed only one PBFMC application. The "license max" value is defined as follows:
· On legacy systems, the upper limit is 1/2 of the maximum number of administrable stations. Legacy platforms do not support SIP trunks.
· On Linux systems, the upper limit is the maximum number of administrable stations.
Maximum Off-PBX Telephones - PVFMC
Number of stations administered for Private Fixed-Mobile Convergence. Each station is allowed only one PVFMC application. The "license max" upper limit is:
· On legacy systems, 1/2 of the maximum number of administrable stations. Legacy platforms do not support SIP trunks.
· On Linux systems, the maximum number of administrable stations.
Maximum Off-PBX Telephones - SCCAN
The "license max" value is defined as follows: · SCCAN is only available on Linux systems. The upper limit is the maximum number of administrable stations.
Maximum Stations
Displays the maximum number of stations allowed in the system.
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System parameters customer options: page 2
Maximum XMOBILE Stations
Specifies the maximum number of allowable XMOBILE stations. In general, each XMOBILE station is assigned to a wireless handset. Each XMOBILE station counts as a station and a port in terms of system configuration.
Platform
Displays the platform being used.
Valid Entry 28 29 30
Usage VCM VCM_ESS VCM_LSP
Platform Maximum Ports
Number of active ports.
Software Package
Indicates whether the software package license is Standard or Enterprise.
Used
The actual current usage as compared to the system maximum.
System parameters customer options: page 2
Maximum Administered Ad-hoc Video Conferencing Ports
Defines the number of ad-hoc ports allowed for the system; one for each simultaneous active conference port. The maximum number of ad-hoc video conferencing ports allowed is the sum of the maximum allowed IP trunks and the maximum allowed SIP trunks on your system.
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System Parameters Customer Options
Maximum Administered IP Trunks
Defines the maximum number of IP trunks administered.
Maximum Administered Remote Office Trunks
Defines the maximum number of IP endpoints based on the endpoint.
Maximum Administered SIP Trunks
Defines the maximum number of SIP trunks administered.
Maximum Concurrently Registered IP eCons
Specifies the maximum number of IP SoftConsoles that can be registered at one time. The maximum number depends on the type of system.
Maximum Concurrently Registered IP Stations
Specifies the maximum number of IP stations that can be registered at one time.
Maximum G700 VAL Sources
Specifies the maximum number of VAL announcement sources.
Maximum Number of DS1 Boards with Echo Cancellation
Displays the number of DS1 circuit packs that can have echo cancellation.
Maximum Video Capable IP Softphones
The maximum number of IP Softphones that are video-capable. The maximum number depends on the type of system.
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System parameters customer options: page 3
Maximum Video Capable Stations
The maximum number of stations that are video-capable. The maximum number depends on the type of system.
Used
For each item with a capacity listed, the USED value is the actual number of units currently in use.
System parameters customer options: page 3
Maximum Number of Avaya Media Server VoIP Channels
This field displays the maximum number of active VoIP channels, which is the number of licenses currently used on active media connections.
Maximum Number of Expanded Meet-me Conference Ports
The maximum number of Expanded Meet-me Conference ports on the system.
Maximum TN2501 VAL Boards
The maximum number of TN2501AP (Voice Announcement over LAN) boards allowed in this system.
Maximum TN2602 Boards with 80 VoIP Channels
The total number of TN2602AP boards that can be administered with 80 VoIP channels.
Maximum TN2602 Boards with 320 VoIP Channels
The total number of TN2602AP boards that can be administered with 320 VoIP channels.
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System Parameters Customer Options
System parameters customer options: page 4
Abbreviated Dialing Enhanced List
Provides the capability to store and retrieve dialing lists that simplify or eliminate dialing. The stored entries are organized in number lists. There are three types of number lists: personal, group, and enhanced.
Access Security Gateway (ASG)
Provides an additional level of security for remote administration.
A/D Grp/Sys List Dialing Start at 01
Used to number Abbreviated Dialing group or system lists starting with 01, rather than simply 1.
Analog Trunk Incoming Call ID
Used to collect and display the name and number of an incoming call on analog trunks.
Answer Supervision by Call Classifier
Indicates if the system can use a call classifier to do answer supervision. If the system has a call classification resource, the field is set to y. The field can be set to y even if the system has no call classifier circuit packs. This circuit pack detects tones and voice-frequency signals on the line and determines whether a call has been answered.
ARS
Provides access to public and private communications networks. Long-distance calls can be routed over the best available and most economical routes. Provides partitioning of ARS routing patterns.
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System parameters customer options: page 4
ARS/AAR Dialing without FAC
Provides for Automatic Alternate Routing (AAR) and Automatic Route Selection (ARS) calls without dialing a feature access code (FAC).
ARS/AAR Partitioning
Provides the ability to partition AAR and ARS into eight user groups within a single server running Avaya Communication Manager. Can establish individual routing treatment for each group.
ASAI Link Core Capabilities
Provides linkage between Avaya Communication Manager and adjuncts. CallVisor ASAI improves the call handling efficiency of ACD agents and other system users by providing an adjunct to monitor, initiate, control, and terminate calls on the server running Communication Manager.
Note: This field applies only to links administered as type asai-ip.
ASAI Link Plus Capabilities
Provides linkage between Avaya Communication Manager and adjuncts. If enabled, then the following ASAI capability groups are also enabled:
· Adjunct Routing · Answering Machine Detection · Selective Listening · Switch Classified Outbound Calls · ISDN Redirecting Number Information - the original dialed number information is provided
within the ASAI messages if it arrives in ISDN SETUP messages from the public networks as either Original Dialed Number or Redirecting Party Number. Note: This field applies only to links administered as type asai-ip.
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System Parameters Customer Options
Asynch. Transfer Mode (ATM) PNC
PNC ATM PNC can be enabled only if: · All prior fiber-link administration has been removed · All "switch-node" and "dup-switch-node" carrier types have been removed.
Asynch. Transfer Mode (ATM) Trunking
If ATM trunking is enabled, multiple ISDN-PRI T1 or E1 trunks can be emulated on one ATM pipe. Enables circuit emulation service (CES). Related links Carrier Medium on page 505
ATMS
Provides for voice and data trunk facilities to be measured for satisfactory transmission performance.
ATM WAN Spare Processor
Indicates whether or not an ATM WAN spare processor is part of the system. An ATM WAN spare processor acts as a PPN in the event of network failure, and can function as an SPE if the main PPN is not functional.
Attendant Vectoring
Enables or disables Attendant Vectoring.
Audible Message Waiting
Enables or disables audible message waiting.
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System parameters customer options: page 4
Authorization Codes
Enables or disables the use of Authorization Codes. Authorization Codes provide levels of calling privileges that override in-place restrictions. In addition to facilities access, authorization codes are used for unique identification for billing security purposes.
CAS Branch
Enables or disables Centralized Attendant Service - Branch.
CAS Main
Enables or disables multi-location customers served by separate switching vehicles to concentrate attendant positions at a single, main Avaya Communication Manager location. The main Avaya Communication Manager is served by an attendant queue that collects calls from all locations (main and branch). Each branch location switches all of its incoming calls to the centralized attendant positions over release link trunks (RLTs). The calls are then extended back to the requested extension at the branch server/switch over the same RLT. When the call is answered, the trunks to the main server are dropped and can be used for another call.
Change COR by FAC
Provides certain users the ability to change the class of restriction of local extensions and local attendants via a telephone by using a feature access code (FAC).
Computer Telephony Adjunct Links
Use this field to administer a Computer Telephony Adjunct (CTI) link between Communication Manager and the adjunct.
Note: This field only applies to links administered as type adj-ip.
Cvg Of Calls Redirected Off-net
Provides continued monitoring for calls redirected to off-network (remote) coverage points. Uses call classification via call classifier circuit pack or ISDN trunk signaling.
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System Parameters Customer Options
DCS (Basic)
Provides transparent operation of selected features across a Distributed Communications System (DCS). Users on one server running Communication Manager can use features located on another server. Includes 4- and 5-digit uniform dialing and 1 to 4 digit steering. Does not support a 6/7-digit dial plan.
DCS Call Coverage
Provides DCS-based transparency of the call coverage feature across a DCS network of media servers or switches.
DCS with Rerouting
Provides for rerouting calls transferred among DCS nodes, enabling rerouting of the call for more effective use of facilities. Related links TSC Supplementary Service Protocol on page 721 Used for DCS on page 953 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836 Group Type on page 708
Digital Loss Plan Modification
Allows or disallows permission to customize the digital loss and digital tone plans.
DS1 MSP
Allows or disallows permission to administer values for the DS1 circuit pack without removing the related translations of all trunks from the trunk group.
DS1 Echo Cancellation
Removes perceivable echo from the system.
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System parameters customer options: page 5
System parameters customer options: page 5
Emergency Access to Attendant
Provides for emergency calls to be placed to an attendant. These calls can be placed automatically by Avaya Communication Manager or dialed by users.
Enable `dadmin' Login
Provides Avaya Business Partners the ability to install, administer, and maintain Avaya servers and switches. The dadmin login has access to all the same commands as other logins with the exception of Go and WP. Go is used for go tcm and go debug as well as go server. WP is for writing memory.
Enhanced Conferencing
Enables or disables the use of Meet-me Conference, Expanded Meet-me Conference, the No Hold Conference features, Selective Conference Party: Display, Drop, or Mute.
Enhanced EC500
Indicates if Extension to Cellular is enabled. EC500 refers to the Extension to Cellular feature.
Enterprise Survivable Server
Identifies the server is a Survivable Core Server (Enterprise Survivable Server).
ESS Administration
Indicates if administration of survivable core server (Enterprise Survivable Servers) is enabled.
Extended Cvg/Fwd Admin
Enables or disables Extended Coverage and Forwarding Administration.
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System Parameters Customer Options
External Device Alarm Admin
Provides for analog line ports to be used for external alarm interfaces. Allows identification of port location, adjunct associated with port location, and the alarm level to report.
Enterprise Wide Licensing
Enterprise Wide Licensing. Go to the Avaya Support website at http://support.avaya.com/ for current documentation, product notices, knowledge articles on Enterprise Wide Licensing.
Five Port Networks Max Per MCC
Allows system administrator to create five port networks in a multi-carrier cabinet. Available only for duplex server Multi-Connect.
Flexible Billing
Provides an internationally accepted standard interface for end-to-end digital connectivity. Used with a T1 interface and supports twenty-three 64-KBPS voice or data B-Channels and one 64Kbps signaling D Channel for total bandwidth of 1.544 Mbps.
Forced Entry of Account Codes
Allows system administration to force account users to enter account codes based on user or trunk class of restriction, or by an option on the Toll Analysis table. FEAC provides an easy method of allocating the costs of specific calls to the correct project, department, and so on.
Global Call Classification
Provides call classification outside of North America. Listens for tones and classifies tones detected. Required for Call Coverage Off Net and Outgoing Call Management.
Hospitality (Basic)
Provides access to basic features including: Attendant Crisis Alert, Attendant Room Status, Automatic Wakeup, Custom Selection of VIP DID Numbers, Do Not Disturb, Names Registration, Single-Digit Dialing, and Mixed Station Numbering.
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System parameters customer options: page 5
Hospitality (G3V3 Enhancements)
Software required for Property Management System and Automatic Wakeup. Property Management System Interface activates Forward PMS Messages to INTUITY Lodging and PMS Protocol Mode (transmit in ASCII mode).
Note: Standard hospitality features are included in basic system software.
IP Attendant Consoles
Controls permission to administer the IP Attendant Console.
IP Stations
Controls permission to administer H.323 and/or SoftPhone stations. Must be enabled for IP telephones.
IP Trunks
Controls permission to administer H.323 trunks. Must be enabled for IP trunks.
ISDN-BRI Trunks
Provides the capability to add ISDN-BRI trunks to Communication Manager. Related links
Service/Feature on page 691 Facility Type on page 630 ISDN-PRI on page 836 TSC Supplementary Service Protocol on page 721 DCS with Rerouting on page 832 Used for DCS on page 953 ISDN-PRI on page 836 Group Type on page 708
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System Parameters Customer Options
ISDN Feature Plus
Provides ISDN Feature Plus signaling.
ISDN-PRI
Provides Integrated Services Digital Network (ISDN-PRI) software for either a switching-hardware platform migration only or a switching-hardware platform migration in combination with a software release upgrade. Also provides signaling support for H.323 signaling. Must be enabled for IP trunks. Related links Service/Feature on page 691 Facility Type on page 630 ISDN-BRI Trunks on page 835 TSC Supplementary Service Protocol on page 721 DCS with Rerouting on page 832 Used for DCS on page 953 ISDN-BRI Trunks on page 835 Group Type on page 708
ISDN/SIP Network Call Redirection
Redirects an incoming ISDN/SIP call from a server running Avaya Communication Manager to another PSTN endpoint. It is used in call centers with Best Service Routing and Lookahead Interflow.
Local Survivable Processor
Indicates that the server is a Survivable Remote Server (Local Survivable Processor). The Survivable Remote Server is configured to provide standby call processing in case the primary media server is unavailable.
Malicious Call Trace
Provides the ability to retrieve certain information related to a malicious call.
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System parameters customer options: page 5
Media Encryption Over IP
Enables or disables media encryption over IP network. Note: For SIP Resiliency feature, ensure that the Media Encryption Over IP field is set to Y.
Mode Code for Centralized Voice Mail
Provides the ability to share a Voice Mail System (VMS) among several servers/ switches using the Mode Code - Voice Mail System Interface.
Multifrequency Signaling
Provides for multi-frequency signaling between Communication Manager and the local telephone company central office.
Multimedia Appl. Server Interface (MASI)
Allows users of the Multimedia Communications Exchange (MMCX) to take advantage of certain Avaya Communication Manager telephony features.
Multimedia Call Handling (Basic)
Allows administration of desktop video-conferencing systems as data modules associated with Avaya Communication Manager voice stations in a multimedia complex. Users can dial one number to reach either endpoint (voice or data) in the complex. Also provides support for IP SoftPhones.
Multimedia Call Handling (Enhanced)
Allows a multifunction telephone to control a multimedia call like a standard voice call.
Multimedia IP SIP Trunking
Extends applicability of the H.323 video station licensing/control to all non-IP Softphones.
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System Parameters Customer Options
System parameters customer options: page 6
Multinational Locations
Provides the ability to use a single Enterprise Communication Server (ECS) with stations, port networks, remote offices, or gateways in multiple countries. Allows administration of location parameters such as companding, loss plans, and tone generation per location, instead of systemwide.
Multiple Level Precedence and Preemption
Multiple Level Precedence and Preemption (MLPP) provides users the ability to assign levels of importance to callers, and when activated, to give higher-priority routing to individual calls based on the level assigned to the caller.
Multiple Locations
Valid Entry y
n
Usage
(Depending on your server configuration, see Avaya Aura® Communication Manager System Capacities Table.) Allows numbering plans, time zone and daylight saving plans, digit analysis, companding, call progress tone generation, loss plan, and analog line board parameters that are specific for each cabinet, Gateway, Remote Office, network regions, signaling groups, and stations. See the Location sections in Avaya Aura®Communication Manager Feature Description, for the other ways, and for a list of features that use location.
The location is fixed. The location is 1 for endpoints and all for digit analysis. This is the default value.
Personal Station Access (PSA)
Provides basic telecommuting package capability for Personal Station Access.
PNC Duplication
Indicates whether or not Port Network Connectivity (PNC) Duplication can be enabled. This feature provides non-standard reliability levels (high, critical, or ATM PNC Network Duplication).
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System parameters customer options: page 6
Port Network Support
Valid Entry y n
Usage The server is operating as a stand-alone Internal Communications Controller (ICC). Traditional Avaya port networks are in use.
Posted Messages
Supports the ability for users to post messages, selected from among a set of as many as 30 (15 fixed, 15 administrable), and shown on display telephones.
Private Networking
Indicates upgrading of PNA or ETN software RTU purchased with earlier systems.
Processor and System MSP
Allows for maintenance of the processor and system circuit packs.
Processor Ethernet
Indicates if the Ethernet card resident in the processor cabinet is used by the Communication Manager Call Processing software in place of a C-LAN card or located in a port network. Only S8300E, HP DL360p G8, HP DL360 G9, Dell R620, and Dell R630 Media Servers display Processor Ethernet Interface. The Processor Ethernet interface is always enabled for duplex media servers.
For the Avaya supported servers, see Avaya Aura® Communication Manager Hardware Description and Reference.
Remote Office
Allows administration of a remote office.
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System Parameters Customer Options
Restrict Call Forward Off Net
Allows the system to monitor the disposition of an off-call and, if it detects busy, bring the call back for further processing, including call coverage.
Secondary Data Module
Provides the ability to use any data module as a secondary data module.
Station and Trunk MSP
Allows for maintenance of the station and trunk circuit packs.
Station as Virtual Extension
Allows multiple virtual extensions to be mapped to a single physical analog telephone. A specific ringing pattern can be administered for each virtual extension. Useful in environments such as college dormitories, where three occupants can have three different extensions for one physical telephone.
System Management Data Transfer
Indicates Communication Manager is accessible by network administration.
Tenant Partitioning
Provides for partitioning of attendant groups and/or stations and trunk groups. Typically this is used for multiple tenants in a building or multiple departments within a company or organization.
Note: If SIP attendant is enabled on console parameters screen, you cannot change the Tenant Partitioning? field value to y.
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System parameters customer options: page 6
Terminal Trans. Init. (TTI)
Allows administrators of Terminal Translation Initialization (TTI) to merge an station administered with X in the Port field, to a valid port by dialing a system-wide TTI security code and the extension from a terminal connected to that port.
Time of Day Routing
Provides AAR and ARS routing of calls based on the time of day and day of the week to take advantage of lower calling rates during specific times. Related links Time of Day Chart on page 129
TN2501 VAL Maximum Capacity
Allows up to 60 minutes storage capacity per pack and multiple integrated announcement circuit packs. This is the Enhanced offer.
Uniform Dialing Plan
Enables or disables three- to seven-digit Uniform Dial Plan (UDP) and one- to seven-digit steering. Also allows use of Extended Trunk Access and Extension Number Portability features.
Usage Allocation Enhancements
Enables the user to use user-defined entries for the incoming and outgoing ISDN calls independent of the NSF value. The Usage Allocation Enhancements feature lets the user administer the non-unique Facility Type and Facility Coding combinations. With Usage Allocation Enhancements, the customer can reserve a minimum number of trunk members for the incoming and outgoing calls at all times.
Note: The Usage Allocation Enhancements feature is helpful outside the US, where the default Network Facilities available in the system are not very useful.
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System Parameters Customer Options
Wideband Switching
Provides wideband data software for switching video or high-speed data. DSO channels can be aggregated up to the capacity of the span. Wideband supports H0, H11, and H12 standards, where applicable, as well as customer-defined data rates.
Wireless
Provides right to use for certain wireless applications.
System parameters customer options: page 7
ACD
Provides the software required for the Call Center Basic, Plus, Deluxe, and Elite features for the number of agents specified. Automatic Call Distribution (ACD) automatically distributes incoming calls to specified splits or skills.
BCMS (Basic)
Provides real-time and historical reports about agent, ACD split, Vector Directory Number (VDN) and trunk group activity.
BCMS/VuStats Service Level
Provides for hunt groups or Vector Directory Numbers (VDNs) with an acceptable service level. An acceptable service level defines the number of seconds within which a call must be answered to be acceptable.
Business Advocate
Enables or disables Avaya Business Advocate. Business Advocate establishes different levels of service for different types of calls. For example, a company might decide that a premium customer gets faster service than other types of customers.
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System parameters customer options: page 7
Call Center Release
The call center release installed on the system.
Call Work Codes
Allows agents to enter digits for an ACD call to record customer-defined events such as account codes or social security numbers.
DTMF Feedback Signals For VRU
Provides support for the use of C and D Tones to voice response units (VRUs).
Dynamic Advocate
Enables or disables the Dynamic Advocate feature. While Business Advocate assigns reserve agents and sets overload thresholds to determine when those reserve agents get engaged, the Dynamic Advocate feature, also known as Dynamic Threshold Adjustment, takes this a step further. Dynamic Advocate automatically adjusts the thresholds as needed to help maintain defined service levels.
EAS-PHD
Increases the number of skills an agent can log in to from 4 to 20. Increases the number of agent skill preference levels from 2 to 16.
Expert Agent Selection (EAS)
Enables or disables skills-based routing of calls to the best-qualified agent.
Forced ACD Calls
See Multiple Call Handling.
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System Parameters Customer Options
Least Occupied Agent
Allows call center calls to be routed to the agent who has been the least busy, regardless of when the agent last answered a call.
Lookahead Interflow (LAI)
Provides Look-Ahead Interflow to balance the load of ACD calls across multiple locations.
Multiple Call Handling (Forced)
Forces an agent to be interrupted with an additional ACD call while active on an ACD call. Splits or skills can be one forced, one per skill, or many forced.
Multiple Call Handling (On Request)
Allows agents to request additional calls when active on a call.
PASTE (Display PBX Data on Phone)
Provides an interface between the display of a DCP telephone set and PC-based applications.
Reason Codes
Allows agents to enter a numeric code that describes their reason for entering the AUX work state or for logging out of the system.
Service Level Maximizer
Allows an administrator to define a service level whereby X% of calls are answered in Y seconds. When Service Level Maximizer (SLM) is active, the software verifies that inbound calls are matched with agents in a way that ensures that the administered service level is met.
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System parameters customer options: page 7
Service Observing (Basic)
Allows a specified user to observe an in-progress call on a listen-only or listen-and-talk basis.
Service Observing (Remote/By FAC)
Allows users to service observe calls from a remote location or a local station using this feature's access codes.
Service Observing (VDNs)
Provides the option of observing and/or monitoring another user's Vector Directory Number (VDN).
Timed ACW
Places an auto-in agent in ACW for an administered length of time after completion of the currently active ACD call.
Vectoring (ANI/II-Digits Routing)
Provides for ANI and II-Digits vector routing used to make vector routing decisions based on caller identity and the originating line.
Vectoring (Basic)
Provides basic call vectoring capability. Related links
Basic on page 93
Vectoring (Best Service Routing)
Enables or disables the Best Service Routing feature. Through special vector commands, Best Service Routing allows the system to compare splits or skills at local and remote locations and queue a call to the resource that will give the caller the best service.
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System Parameters Customer Options
Vectoring (CINFO)
Enables or disables the Caller Information Forwarding (CINFO) feature that allows the collection of caller-entered digits (ced) and customer database provided digits (cdpd) for a call from the network. Related links CINFO on page 93
Vectoring (G3V4 Advanced Routing)
Provides for Rolling Average Speed of Answer Routing, Expected Wait Time Routing, and VDN Calls Routing.
Vectoring (G3V4 Enhanced)
Allows the use of enhanced comparators, wildcards in digit strings for matching on collected digits and ANI or II-digits, use of Vector Routing Tables, multiple audio/music sources for use with waittime command and priority level with the oldest-call-wait conditional.
Vectoring (Holidays)
Indicates if the Holiday Vectoring feature is enabled or disabled that simplifies vector writing for holidays.
Vectoring (Prompting)
Allows flexible handling of incoming calls based on information collected from the calling party or from an ISDN-PRI message.
System parameters customer options: page 8
Logged-In ACD Agents
The total number of ACD agents that can be logged in simultaneously.
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System parameters customer options: page 8
The limit applies to ACD agents on ACD and EAS calls. Auto-Available Split (AAS) agent ports are counted when they are assigned. AAS split or skill members are also counted. If the port for an AAS split/skill member is logged out, (for example, when a ringing call is redirected) the logged-in agent count is not updated. These counts are updated only during administration.
Logged-In Advocate Agents
The total number of Business Advocate Agents logged in simultaneously. The number of logged-in Business Advocate agents counts towards the total number of logged-in ACD agents.
Logged-In IP Softphone Agents
The total number of IP Softphone agents that can be logged-in simultaneously.
VDN of Origin Announcement
Provides a short voice message to an agent indicating the city of origin of the caller or the service requested by the caller based on the VDN used to process the call.
VDN Return Destination
Allows an incoming trunk call to be placed back in vector processing after all parties, except the originator, drop.
VuStats
Puts call center statistics on agents, splits or skills, Vector Directory Numbers (VDNs), and trunk groups on telephone displays.
VuStats (G3V4 Enhanced)
Provides G3V4 VuStats enhancements including historical data and thresholds.
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System Parameters Customer Options
System parameters customer options: page 9 (QSIG OPTIONAL FEATURES)
Basic Call Setup
Provides basic QSIG services: basic connectivity and calling line ID number. Either ISDN-PRI or ISDN-BRI Trunks must be enabled for the system. Related links ISDN-PRI on page 836 ISDN-BRI Trunks on page 835
Basic Supplementary Services
Provides the following QSIG Supplementary Services: · Name ID · Transit Capabilities; that is, the ability to tandem QSIG information elements · Support of Notification Information Elements for interworking between QSIG and non-QSIG tandemed connections · Call Forwarding (Diversion) by forward switching. No reroute capabilities are provided · Call Transfer by join. No path replacement capabilities are provided. · Call Completion (also known as Automatic Callback)
Either ISDN-PRI or ISDN-BRI Trunks must be enabled for the system. Related links
ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
Centralized Attendant
Allows all attendants in one location to serve users in multi locations. All signaling is done over QSIG ISDN lines.
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System parameters customer options: page 10 (ASAI FEATURES)
Interworking with DCS
Allows the following features to work between a user on a DCS-enabled media server or switch in a network and a QSIG-enabled media server or switch:
· Calling/Called/Busy/Connected Name · Voice Mail/Message Waiting · Leave Word Calling
Supplementary Services with Rerouting
Provides the following QSIG Supplementary Services: · Transit Capabilities; that is, the ability to tandem QSIG information elements. · Support of Notification Information Elements for interworking between QSIG and non-QSIG tandemed connections. · Call Forwarding (Diversion) by forward switching. In addition, reroute capabilities are provided. · Call Transfer by join. In addition, path replacement capabilities are provided.
Transfer Into QSIG Voice Mail
Allows transfer directly into the voice mail box on the voice mail system when a QSIG link connects Avaya Communication Manager and the voice mail system.
Value Added (VALU)
Provides additional QSIG functionality, including the ability to send and display calling party information during call alerting.
System parameters customer options: page 10 (ASAI FEATURES)
Agent States
This field provides proprietary information that is used by Avaya applications.
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System Parameters Customer Options
Note: This field applies only to the CTI links administered as type adj-ip. Go to the Avaya Support site. http://support.avaya.com/ for current documentation, product notices, and knowledge articles on proprietary information used by Avaya applications,
CTI Stations
Use the CTI Stations field to enable or disable any application using a CTI link and a CTI station to receive calls.
Phantom Calls
Indicates if phantom calls are enabled. This field only applies to links administered as type ADJIP or ASAI-IP.
System parameters customer options: page 11 (MAXIMUM IP REGISTRATIONS BY PRODUCT ID)
Limit
Valid Entry 1000 5000
Usage Maximum number of IP registrations allowed, depending on server configuration.
Product ID
Identifies the product using the IP (internet protocol) registration.
These are just a few examples of valid Product IDs. The valid Product IDs for a system are controlled by the license file.
Product ID example
IP_API_A
Description Application Enablement Services
Table continues...
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System parameters customer options: page 11 (MAXIMUM IP REGISTRATIONS BY PRODUCT ID)
Product ID example IP_Agent IP_eCons IP_Phone IP_Soft oneX_Comm
Description
One-X agent Soft Console IP attendant IP telephones IP softphones One-X Communicator
Rel
Valid Entry *
Usage IP endpoint can register with any release for the product ID.
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Chapter 129: System parameters Duplication
Use the System Parameters Duplication screen to enable PNC or IPSI duplication. Example command: change system-parameters duplication
Enable Operation of IPSI Duplication
Enables or disables IPSI duplication.
Enable Operation of PNC Duplication
Enables or disables PNC duplication. Available only if PNC Duplication is enabled for the system. Related links
PNC Duplication on page 838
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Chapter 130: System parameters maintenance
This screen is described in Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and Servers.
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Chapter 131: System Parameters Media Gateway Automatic Recovery Rule
This screen defines rules for returning a fragmented network, where a number of Branch Gateways are being serviced by one or more Survivable Remote Server (Local Survivable Processors) automatically to the primary media server. The system displays a different warning message or time window grid depending on the option selected for the Migrate H.248 MG to primary field. In the time window grid, an x or X can be specified for each hour. Leave blank to disable. This method helps with overlapping time issues between days of the week. There is no limit on intervals. For more information on Auto Fallback for Branch Gateways, see Administering Network Connectivity on Avaya Aura® Communication Manager. Example command: change system-parameters mg-recovery-rule n, where n is the assigned recovery rule number.
Migrate H.248 MG to primary
Use this field to indicate auto-fallback preferences.
Valid entry immediately
0-active calls time-day-window
time-window-OR-0 -active-calls
Usage
The system accepts the first gateway registration that comes from the gateway, regardless of call count or time of day. This is the default.
The system accepts the first gateway registration reporting a 0-active call count.
The system accepts a valid gateway registration message received during any part of the interval. When you select this option, the system displays a grid for defining required hours or days for the time window.
The system accepts a valid gateway registration reporting a 0-active call count or a valid registration with any call count is received during the specified time-day intervals. When you select this option, the system displays a grid for defining required hours or days for the time window.
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Minimum time of network stability
Minimum time of network stability
Valid Entry 3 to 15
Usage
Administers the time interval for stability in the H.248 link before auto-fallback is attempted. Default is 3.
Recovery Rule Number
The number of the recovery rule up to the server maximum.
Rule Name
The name for the recovery rule in alpha-numeric characters. The recovery rule is an aid for associating rules with gateways.
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Chapter 132: Media Server Recovery Rules
Communication Manager is now integrated with Avaya Aura® Media Server (MS), a software-based media application platform where all the media is processed on software, not on dedicated hardware. Avaya Aura MS is another type of media resource provider besides Port Networks (PN) and Media Gateways (MG).
Communication Manager sends a status message to each Avaya Aura® MS on the Communication Manager reporting list. The status message indicates whether Communication Manager is active or inactive for the specific media server.
Avaya Aura® MS serves both as a media server and a status keeper. Each server, whether the main server or the survivable server, sends the status information to Avaya Aura® MS and requests status information about other servers. Avaya Aura® MS is a conduit that transmits the information to Communication Manager when requested. Based on the status information received, each server processes survivability actions.
Type the command change system-parameters ms-recovery-rule.
Auto Return
Use the Auto Return field to control the recovery or fallback to the main server.
Valid entry yes no
scheduled
Usage
The fallback to the main server is automatically done as soon as possible, after the specified time delay.
The fallback to the main server is manual. You must type the command enable ms-return.
If you prefer to start the recovery process in off-peak hours, you can set the field to no.
The fallback to the main server is scheduled for the specified day and time.
The recovery occurs at the next scheduled occurrence.
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Day
Day
Use the Day field to start the recovery of the main server on the specified day. The Day field is available only when you set the Auto Return field to scheduled. You can set any day of the week to start the recovery.
Report Expiration (sec)
Use the Report Expiration field to set the duration for Avaya Aura® MS to retain the status message of Communication Manager. A short report expiration facilitates a rapid failover. However, the expiration increases the risk of unintended failover due to temporary conditions, such as a server reboot. You must type a number in seconds from 2 through 400. The default value is 180 seconds.
Report Interval (sec)
At each report interval, Communication Manager sends a status message to each Avaya Aura® MS on the Communication Manager reporting list. The status message indicates whether Communication Manager is active or inactive for the specific media server. A short report interval increases network traffic and the CPU load of Communication Manager and Avaya Aura® MS. You must type a number in seconds from 1 through 300. The default value is 60 seconds.
Time
Use the Time field to start the recovery of the main server at the specified time. The Time field is available only when you set the Auto Return field to scheduled. You must type a number from 0 through 23 for hours and from 0 through 59 for minutes. Do not leave the field blank. Type values in both the hour and minute fields.
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Media Server Recovery Rules
Time Delay
Use the Time Delay field to set the duration for the main server to connect to Avaya Aura® Media Server. The connection to the main server is established only when the network is stable enough to start the fallback. The Time Delay field is available only when you set the Auto Return field to yes. You must type a number in minutes from 0 through 120. The default value is 0. Do not leave the field blank.
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Chapter 133: System parameters OCM call classification
Administers the country tone characteristics for Outbound Call Management (OCM) applications. It is not required for United States OCM applications. This screen defines the busy tone and cadence and can be administered with up to four on and off steps, which is four valid cycles to determine busy tone. Use a minimum of two on and off steps to determine a valid busy tone. If the cadence is administered with one on and off step, any time the classifier hears the cadence it is BTD signal. Available only if Global Call Classification is enabled for the system, or when Enable Busy Tone Disconnect for Analog loop-start Trunks is enabled for the system. Example command: change system-parameters ocm-call-classification
Related links Global Call Classification on page 834 Enable Busy Tone Disconnect for Analog Loop-start Trunks on page 821
System parameters OCM call classification: page 1
Cadence Classification After Answer
Valid Entry y
n
Usage
Communication Manager puts call classifiers into modes that listen for call progress tones while listening the voice, with or without Answering Machine Detection (AMD).
No-cadence call classification modes detect live voice with or without AMD whenever Communication Manager receives a connect or answer supervision message from an outbound trunk. These modes do not detect any call progress tone cadence.
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System parameters OCM call classification
Global Classifier Adjustment (dB)
Valid Entry 0 to 15
Usage
Specifies the dB loss adjustment. where 0 is the least adjustment, and 15 is the most adjustment.
USA Default Algorithm
Enables or disables the use of the default United States tone detection.
USA SIT Algorithm
Valid Entry y n
Usage
Uses the United States (SIT) tone characteristics for SIT tone detection.
The system treats tones with the administered tone name intercept as if they were SIT VACANT, and treats tones with the administered tone name information as if they were SIT UNKNOWN.
System parameters OCM call classification: page 2
Cadence Step
Identifies the number of each tone cadence step and indicates whether the tone is on or off during this cadence step.
Duration Maximum
Valid Entry 75 to 6375
Usage The upper limit in milliseconds of the tone duration in increments of 25.
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System parameters OCM call classification: page 2
Duration Minimum
Valid Entry 75 to 6375
Usage The lower limit in milliseconds (msec) of the tone duration in increments of 25.
Instance
Valid Entry 1 to 8
Usage
The number distinguishes tones that have the same name but more than one definition of silence and tone-on characteristics.
Tone Continuous
Valid Entry y n
Usage Indicates a continuous tone. Indicates a non-continuous tone.
Tone Name
Valid Entry busy information intercept reorder ringback
Usage
The name of the tone. Busy is required when Busy Tone Disconnect for Analog Loop-start Trunks is enabled or Global Call Classification is disabled for the system.
Required for tone definition outside of the U.S. and Canada.
Related links Global Call Classification on page 834 Enable Busy Tone Disconnect for Analog Loop-start Trunks on page 821
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Chapter 134: System Parameters Port Networks
These screens assign port networks to communities and specify recovery rules for port networks to return to the main server. Example command: change system-parameters port-networks
System parameters port networks: page 1
Community
Valid Entry 1 to 64
Usage The Network Community number associated with this port network.
Note:
If the port network is administered in the system, the default community is 1 and administrable with a value between 1 and 64. If the port network is not administered in the system, the community value is 1 and not administrable.
PN
Valid Entry 1 to 64
Usage Displays the port network.
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System parameters port networks: page 2
System parameters port networks: page 2
Auto Return
The Auto Return functionality is used to schedule a day and time for all port networks to return to the control of the main server after a failover occurs. The schedule can be set up to seven days prior the its activation.
Valid Entry y
n
s(cheduled)
Usage
Enables the Auto Return feature. The port networks can automatically return to the main server after the value set in the IPSI Connection up time expires.
Disables the Auto Return feature. The port networks cannot automatically return to the control of the main server.
Enables the Auto Return feature. A day and time also needs to be administered to schedule a day and time to return the port networks to the control of the main server.
Day
Valid Entry Monday through Sunday
Usage
The day of the week the port networks return to the control of the main server if Auto Return is administered on a schedule.
IPSI Connection Up Time (min)
Valid Entry 3 to 120
Usage
The number of minutes that the IP Server Interface (IPSI) waits to return to the main server after communication with the main server is restored.
No Service Time Out Interval
Valid Entry 3 to 15
Usage
The time in minutes that the IP Server Interfaces (IPSIs) wait before requesting service from the highest survivable core server on its priority list. Default is 5 minutes.
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System Parameters Port Networks
PN Cold Reset Delay Timer (sec)
Valid Entry 60 to 120
Usage
Time in seconds before a PN cold reset occurs. This value is retained after an upgrade event. Default is 60 seconds.
Time
Valid Entry 00:00 to 23:59
Usage
The time of day the port networks return to the control of the main server if Auto Return is administered on a schedule. This field uses a 24-hour military format.
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Chapter 135: System Parameters - SCCAN
Example command: change system-parameters sccan
Announcement
The extension of the announcement played during call hand-in or handout.
H1 Handover
The primary handover extension called to facilitate handover of a cellular call to the WAN or WLAN. Depending on whether the user is entering or exiting the Enterprise space, Communication Manager replaces the active call with the new call made using the hand-off H1 or H2 number.
H2 Handover
A secondary handover extension used when no acknowledgement is received from the H1 Handover number.
MM (WSM) Route Pattern
A route pattern number that is SCCAN-enabled. Partition route pattern indexes, RHNPA indexes, deny, or nodes are not supported. If this field is left blank, the feature is turned off. Blank is the default.
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System Parameters - SCCAN
Special Digit Conversion
Valid Entry y
n
Usage
Allows a user to call a cellular telephone number and get the same treatment as calling an extension that is running Communication Manager. ARS checks the dialed string to determine if the dialed string is a SCCAN telephone number. If the number is a SCCAN telephone number, the cellular telephone number is replaced with the extension number that the cellular telephone is mapped to.
The feature is turned off. This is the default.
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Chapter 136: System Parameters Special Applications
Use this screen to enable the Special Application features. A Special Application feature is a custom feature requested by the customer for a specific need. Special Application features are included in the Communication Manager software base, however additional administration is required. For more information about the Special Application features, see Avaya Aura® Communication Manager Special Application Features.
Note: · Communication Manager provides the UCID of the related held call ID to the new call ID when any one of the following conditions are satisfied: - Communication Manager receives the third-party Auto Dial request and the specialapplication SA8348 is selected as 'y'. - Communication Manager receives the third-party Auto Dial request and the specialapplication SA8702 and the field "Copy UCID for Station Conference/Transfer?" are both selected as `y'. · Communication Manager provides the ASAI-UUI of the related held call ID to the new call ID when: - Communication Manager receives the third-party Auto Dial request and the field "Copy ASAI UUI During Conference/Transfer?" is selected as `y'.
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Chapter 137: Telecommuting Access
Administers the extension which allows remote users to use the Telecommuting Access feature. Example command: change telecommuting-access
Telecommuting Access Extension
An extension that allows only remote access to the Telecommuting Access feature. This extension must consist of one to 16 digits that conforms to the system dial plan and is not assigned to any other system object.
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Chapter 138: Tenant
This screen defines tenants to the system. If your server running Communication Manager uses tenant partitioning, see Tenant Partitioning in Avaya Aura® Communication Manager Feature Description and Implementation. Example command: change tenant n, where n is the tenant partitioning number.
Tenant: page 1
Attendant Group
Valid Entry digits
Usage
The attendant group assigned to the tenant. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
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Tenant
Attendant Vectoring VDN
Assigns the VDN extension for Attendant Vectoring to a console. Available only if Attendant Vectoring is enabled for the system and Tenant Partitioning is disabled.
Related links Attendant Vectoring on page 830 Tenant Partitioning on page 840
COS Group
Valid Entry digits
Usage
The Class of Service (COS) group assigned to the tenant. Available only if Tenant Partitioning is enabled for the system. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Related links Tenant Partitioning on page 840
Ext Alert Port (TAAS)
The Trunk Answer Any Station (TAAS) alert port assigned to this tenant, if any. The port type and the object type must be consistent. The port can be assigned to only one tenant.
Valid Entry A valid port address or X
1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks)
01 to 31 1 to 250
Usage First and second characters are the cabinet number
Third character is the carrier Fourth and fifth character are the slot number Six and seventh characters are the circuit number
Gateway
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Tenant: page 1
Ext Alert (TAAS) Extension
The extension of the external alert TAAS extension. A system installer can then use the Terminal Translation Initialization (TTI) feature from a telephone plugged into any port to assign this extension number to that port. Doing so makes that port the external alert TAAS port.
Available only if Ext Alert Port (TAAS) is administered without hardware.
Related links Ext Alert Port (TAAS) on page 870
Music Source
Valid Entry digits
Usage
The music or tone source for this tenant partition. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Night Destination
The night service station extension, if night service is wanted for this tenant.
SIP
Use this field to assign a SIP attendant to a tenant without using SIP stations.
Note:
If native attendant like type "302" or "console" is already configured in the same Attendant Group, then the Communication Manager will not allow you to enable SIP on Tenant form. Also, if SIP is enabled on the Tenant form, Communication Manager will not allow you to add a new attendant user of type "302" or "console" in the same Attendant Group. Before SIP is enabled, Attendant Vectoring VDN field must be blank.
Valid Entry n y
Usage Disables SIP attendant. This is the default value. Enables SIP attendant. When you change the value to y, then the following values appear:
· Routing Digits
· Attendant Server
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Tenant
Routing Digits Access code to route calls to SIP attendant server. You can enter up to four digits only. The Routing Digits field appears on the Tenant form, only if the Tenant Partition is enabled.
Valid Entry digits from 0 to 9
blank
Usage
Digits comprising the AAR or ARS access code. The default value is blank.
Call routing to attendant server is not specified.
Attendant Server Attendant server number. This field cannot be blank. The Attendant Server field appears on the Tenant form, only if the Tenant Partition is enabled.
Valid Entry digits from 0 to 9
Usage You can enter up to 17 digits.
Tenant
Displays the tenant number.
Tenant Description
A description of the tenant that consists of up to 40 alpha-numeric characters.
DISTINCTIVE AUDIBLE ALERTING
Available only if Tenant Partitioning is enabled for the system.
Attendant Originated Calls
Indicates which type of ringing applies to attendant-originated calls.
Valid Entry internal external priority
Usage Internal ringing applies to attendant-originated calls. External ringing applies to attendant-originated calls. Default is external. Priority ringing applies to attendant-originated calls.
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Tenant: page 2
Distinctive Audible Alerting (Internal, External, Priority)
The number of rings for Internal, External, and Priority calls. For virtual stations, this applies to the mapped-to physical telephone. This is also known as Distinctive Ringing. Defaults are as follows:
1. Internal calls
2. External and attendant calls
3. Priority calls
Note:
SIP messaging includes the ring types internal, external, intercom, auto-callback, hold recall, transfer recall, or priority. In Communication Manager, types intercom, autocallback, hold recall, and transfer recall are treated as priority.
Valid Entry 1 2 3
Usage 1 burst, meaning one burst of ringing signal per period 2 bursts, meaning two bursts of ringing signal per period 3 bursts, meaning two bursts of ringing signal per period
Related links Distinctive Audible Alerting (Internal, External, Priority) on page 311
Tenant: page 2
This screen is used to enable additional tenant-to-tenant calling permissions.
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Tenant
Calling permissions
Establishes or blocks calling permission between the tenant being administered and any other tenant. The system default allows each tenant to call only itself and Tenant 1.
Tenant
Displays the tenant number.
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Chapter 139: Terminal Parameters
Administers system-level parameters and audio levels for the 603 CALLMASTER telephones and the 4600-series, 6400-series, 8403, 8405B, 8405B+, 8405D, 8405D+, 8410B, 8410D, 8411B, 8411D, 8434D, and 2420/2410 telephones. Only authorized Avaya personnel can administer this screen.
Note: With the Multinational Locations feature enabled, terminal parameters can be administered per location, rather than system-wide. Example command: change terminal-parameters
Base Parameter Set
Determines which default set of telephone options and levels will be used. This field corresponds to the country codes. Related links Country options table on page 817
Customize Parameters
Allows or disallows permission for the administrator to change one or more of the default parameters.
ADJUNCT LEVELS
Touch Tone Sidetone (dB)
Determines the touchtone volume fed back from the telephone when a user presses a button.
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Terminal Parameters
Voice Receive (dB)
Determines the volume of voice inbound to the adjunct.
Voice Sidetone (dB)
Determines the volume of voice fed back from the handset voice microphone to the user's ear.
Voice Transmit (dB)
Determines the volume of voice outbound from the adjunct.
OPTIONS
Display Mode
Determines how the # and ~ characters appear on the telephone's display.
Valid Entry 1 2
Usage
The # and ~ do not change.
The telephone displays a # as a British pound sterling symbol and a ~ as a straight overbar.
DLI Voltage Level
Determines whether DCP Line Voltage used by the telephones is forced high, forced low, or allowed to automatically adjust.
Handset Expander Enabled
Determines whether or not the telephone reduces background noise on the handset.
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OPTIONS
Volume for DCP Types
Determines what volume adjustments are retained. Allows the DCP telephone volume to be adjusted while the call is in-progress.
Valid Entry
default speaker, handset unchangeable
default settings used to begin each call
retain handset and speaker between calls
retain speaker, handset unchangeable
Usage The speaker resets to the default settings while the adjusted handset setting is retained. No adjusted handset and speaker settings are retained.
The adjusted handset and speaker settings are retained
Only the adjusted speaker setting is retained.
Volume for IP Types
Determines what volume adjustments are retained. Allows the IP telephone volume to be adjusted while the call is in progress.
Note:
Avaya recommends that values for PRIMARY LEVELS or BUILT-IN SPEAKER LEVELS are not changed.
Valid Entry
default speaker, handset unchangeable
default settings used to begin each call
retain handset and speaker between calls
retain speaker, handset unchangeable
Usage The speaker resets to the default settings while the adjusted handset setting is retained. No adjusted handset and speaker settings are retained
The adjusted handset and speaker settings are retained
Only the adjusted speaker setting is retained
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Terminal Parameters
Touch Tone Transmit (dB)
Determines the touchtone volume fed outbound from the telephone.
Voice Receive (dB)
Determines the volume of voice inbound to the adjunct.
Voice Transmit (dB)
Determines the volume of voice outbound from the adjunct.
PRIMARY LEVELS
The system displays the default setting from the Base Parameter Set for all fields. Also, these fields all require the same input; valid entries are from -44.0 db through +14.0 db in 0.5 increments. For example, -44.0, -43.5, -43.0 and so on.
Touch Tone Sidetone (dB)
Determines the touchtone volume fed back from the telephone when a user presses a button.
Touch Tone Transmit (dB)
Determines the touchtone volume fed outbound from the telephone.
Voice Receive (dB)
Determines the volume of voice inbound to the adjunct.
Voice Sidetone (dB)
Determines the volume of voice fed back from the handset voice microphone to the user's ear.
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Voice Transmit (dB)
Determines the volume of voice outbound from the adjunct.
PRIMARY LEVELS
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Chapter 140: Terminating Extension Group
Defines a Terminating Extension Group (TEG). Any telephone can be assigned as a TEG member; however, only a multi-appearance telephone can be assigned a TEG button with associated status lamp. The TEG button allows the telephone user to select a TEG call appearance for answering or for bridging onto an existing call. The TEG members are assigned on an extension number basis. Call reception restrictions applicable to the group are specified by the group class of restriction (COR). The group COR takes precedence over an individual member's COR. The members could all be termination restricted but still receive calls if the group is not restricted. An extension number can be assigned to more than one TEG but can have only one appearance of each group. Example command: change term-ext-group n, where n is the assigned group number.
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AUDIX Name
AUDIX Name
The name of the AUDIX machine. Must be the same name as the IP Node name and administered after the IP Node is configured. Related links IP Node Names on page 475 IP Node Names on page 475
COR
Valid Entry 0 to 995
Usage The class of restriction (COR) number that reflects the required restrictions.
Coverage Path
The call coverage path number for this group. A TEG cannot serve as a coverage point; however, calls to a TEG can redirect to coverage.
Valid Entry digits
t1 to t999 blank
Usage Path number. For the maximum allowed entry, see Avaya Aura® Communication Manager System Capacities Table.
Time of day table
Not administered
Group Extension
The extension of the terminating extension group. The extension can consist of one to seven digits and cannot be a Vector Directory Number (VDN). This field cannot be left blank.
Group Name
The name used to identify the terminating extension group.
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Terminating Extension Group
Group Number
Displays the terminating extension group number.
ISDN Caller Disp
Specifies whether the TEG group name or member name (member of TEG where call terminated) is sent to the originating user. Required if ISDN-PRI or ISDN-BRI Trunks are enabled for the system.
Valid Entry grp-name mbr-name blank
Usage TEG group name is sent. Member name is sent. Not administered.
Related links ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
LWC Reception
Use this field to specify the location where the system must store the LWC messages.
Valid entry spe none audix
Usage Use this option to store the LWC messages on Switch Processor Element (SPE). Use this option if you do not want to store the LWC messages. Use this option to store the LWC messages on the voice messaging system.
Related links AUDIX Name on page 411
Security Code
Use this field to enter the four-digit station security code for the Demand Print messages feature.
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TN
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
GROUP MEMBER ASSIGNMENTS
Ext
The extension number of one to seven digits assigned to a group number. The extension cannot be a Vector Directory Number (VDN).
Name
Displays the name assigned to the group number.
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Chapter 141: TFTP Server
The Trivial File Transfer Protocol screen allows specification of the TFTP server that Avaya Communication Manager uses to get download files. Example command: change tftp-server
Filename in Memory
Displays the name of the file currently in Communication Manager memory.
File Size
Displays the number of bytes transferred.
File Status
Displays the download status as Download In Progress, Download Failed, File Not Found, or Download Completed.
File to Retrieve
The name of the file to retrieve using up to 32 alphanumeric, case sensitive, characters for identification.
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Local Node Name
Local Node Name
A previously-administered local node name. The node must be assigned to a CLAN IP interface or procr (processor CLAN).
Valid Entry 1 to 15 procr
Usage The node name. Processor CLAN for S8300E Server
Related links Name on page 475
Station Type
Displays the information related to the firmware, font, language, boot, and Digital Signal Processor (DSP) when a firmware, language file, font, DSP, or boot loader file download successfully.
TFTP Server Node Name
A previously-administered TFTP server node name from 1 to 15 characters. Related links
Name on page 475
TFTP Server Port
Valid Entry 1 to 64500
Usage The number for the remote TCP port.
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Chapter 142: Time of Day Coverage Table
This screen allows administration of up to five different coverage paths, associated with five different time ranges, for each day of the week. Only one coverage path can be in effect at any one time. Example command: change coverage time-of-day n, where n is the assigned time of day coverage table.
Act Time
Valid Entry 00:01 23:59
Usage
Specifies the activation time of the associated coverage path. Information must be entered in 24-hour time format.
If there are time gaps in the table, there will be no coverage path in effect during those periods. The first activation time for a day is set to 00:00 and cannot be changed. Activation times for a day must be in ascending order from left to right.
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CVG Path
Valid Entry 1 to 9999 blank
Usage The coverage path number.
Time of Day Coverage Table
Displays the Time of Day Coverage Table number.
CVG Path
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Chapter 143: Time of Day Routing Plan
This screen is used to set up Time of Day Routing Plans.
· AAR and ARS calls can be routed based on the time of day each call is made. · Up to eight Time of Day Routing Plans can be designed, each scheduled to change up to six
times a day for each day in the week. · The Time of Day Routing Plan PGN# is matched with the PGN# field on the Partition Routing
Table for the correct route pattern.
Available only if Automatic Route Selection (ARS) or Private Networking, AAR/ARS Partitioning, and Time of Day Routing are enabled for the system.
Example command: change time-of-day n, where n is the time of day routing plans.
Related links PGN 1 (through PGN 8) on page 640 ARS on page 828 Private Networking on page 839 Time of Day Routing on page 841
Act Time
Valid Entry 00:00 to 23:59
Usage
Specifies the time of day the route pattern, identified by PGN#, begins. Time is represented using a 24 hour clock. List times for the same day in increasing order. There must be at least one entry for each day.
PGN #
The route pattern for the activation time listed. The PGN must match a previously-administered PGN and route pattern on the Partition Routing Table. There must be at least one entry for each day.
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Valid Entry 1 to 8 1 to 32
Time of Day Routing Plan
Usage The route pattern range when SA9050 is inactive. The route pattern range when SA9050 is active. The range depends on the type of Communication Manager template. The default value is 8. (For more information on how SA9050 changes capacities, see Avaya Aura® Communication Manager System Capacities Table.)
Time of Day Routing Plan
Displays the Time of Day Routing Plan number.
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Chapter 144: Time of Day Station Lock Table
This screen administers the ability to lock stations automatically by a time of day schedule. Example command: change tod-station-lock n, where n is the assigned time of day station lock table number.
Begin Time
Valid Entry 00:00 to 23:59
Usage
When the time of day station lock interval begins for each day of the week. Time is represented using a 24-hour clock.
End Time
Valid Entry 00:00 to 23:59
Usage
When the time of day station lock interval ends for each day of the week. Time is represented using a 24-hour clock.
INTERVAL 1, 2, 3
Three separate time of day station lock intervals can be administered. The system imposes validation of overlapping intervals or invalid blank entries.
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Manual Unlock allowed
Manual Unlock allowed
Indicates whether or not the user can manually unlock the TOD-locked station using either a stalock button or a Feature Access Code followed by an SSC.
Table Active
Activates or deactivates time of day lock for all stations associated with this table.
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Chapter 145: Toll Analysis
This screen associates dialed strings to the system's Restricted Call List (RCL), Unrestricted Call List (UCL), and Toll List. Users can be forced to dial an account code if you associate dialed strings with CDR Forced Entry of Account Codes. To maximize system security, toll calling areas must be restricted as much as possible through the use of the RCL (Restricted Call List) and Toll List fields on this screen.
Note: The Toll List field on this screen does not interact with or relate to the ARS Toll Table. Example command: change toll n, where n is the dialed digits.
CDR FEAC
Valid Entry x
Usage
Requires an account code from a call whose facility COR requires a Forced Entry of Account Code.
Dialed String
Communication Manager matches the dialed numbers with the entry in the Dialed String field that most closely matches the dialed number. You can enter up to 18 digits that the call-processing server analyzes. You can also enter the wildcard characters, x and X.
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Location
Location
Valid Entry
1 to 250 or 1 to 2000
Depending on your server configuration, see Avaya Aura® Communication Manager System Capacities Table.
all
Usage The location of the endpoint that is dialing the digits. Available only if ARS and Multiple Locations are enabled for the system. See the Location sections in Avaya Aura® Communication Manager Feature Description and Implementation, for the other ways to assign a location, and for a list of features that use location.
Indicates that this Toll Analysis Table is the default for all locations.
Related links ARS on page 828 Multiple Locations on page 838
Max
Use this field to specify the maximum number of user-dialed digits that the system must collect to match the dialed string.
Min
Use this field to specify the minimum number of user-dialed digits that the system must collect to match the dialed string.
Percent Full
This field displays the percentage of system memory resources used for the AAR and the ARS digit analysis. For more information about the system memory resources, see Avaya Aura® Communication Manager System Capacities Table.
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Toll Analysis
RCL
Valid Entry x
Usage
Assigns the Dialed String to the Restricted Call List (RCL). All entries marked with x and their associated dialed strings are referred to as the System's Restricted Call List. The RCL can be assigned to any COR.
Toll List
Valid Entry x
Dialed String 0 1 20 21 30 31 40 41 50 51 60 70 71 80 81 90 91
Usage Assigns the Dialed String to the Toll List.
Minimum 1 4 10 10 10 10 10 10 10 10 10 10 10 10 10 10 10
Maximum 23 23 10 10 10 10 10 10 10 10 10 10 10 10 10 10 10
Toll List X X X X X X X X X X X X X X X X X
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Unrestricted Call List
Unrestricted Call List
Valid Entry x
Usage Assigns the dialed string to one of the system's Unrestricted Call Lists (UCL).
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Chapter 146: Tone Generation
This screen administers the tone characteristics that parties on a call hear under various circumstances.
Note: With the Multinational Locations feature enabled, tone generation can be administered per location, rather than system-wide.
Tone Generation: page 1
440Hz PBX-dial Tone
Valid Entry y n
Usage The switch primary dial tone is changed to a continuous 440Hz-17 tone. A customized tone defaults to the Base Tone Generation Set.
Related links Base Tone Generator Set on page 897
440Hz Secondary-dial Tone
Valid Entry y
n
Usage
The secondary local telephone company central office dial tone is changed to a continuous 440Hz-17 tone.
A customized tone defaults to the Base Tone Generation Set.
Related links Base Tone Generator Set on page 897
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Tone generation: page 2 (TONE GENERATION CUSTOMIZED TONES page)
Base Tone Generator Set
Valid Entry 1 to 50
Usage The country code that identifies the base tone generation set to be used.
Related links Country options table on page 817
Tone generation: page 2 (TONE GENERATION CUSTOMIZED TONES page)
Cadence Step
Displays the number of each tone cadence step.
Duration (msec)
Valid Entry 50 to 12750
Usage The duration of this step in the tone sequence. Values are in increments of 50.
Step
The number of the cadence step for this goto command.
Tone (Frequency/Level)
Valid Entry silence
goto
Usage No tone. A final step of silence with an infinite duration will be added internally to any tone sequence that does not end in a goto. Repeats all or part of the sequence, beginning at the specified cadence step.
Table continues...
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Tone Generation
Valid Entry 350/-17.25 350+425/-4.0 350+440/-13.75 375+425/-15.0 404/-11.0 404/-16.0 404+425/-11.0 404+450/-11.0 425/-4.0 425/-11.0 425/-17.25 440/-17.25 440+480/-19.0 450/-10 480/-17.25 480+620/-24.0 525/-11.0 620/-17.25 697/-8.5 770/-8.5 852/-8.5 941/-8.5 1000/0.0 1000/+3.0 1004/0.0 1004/-16.0 1209/-7.5 1336/-7.5 1400/-11.0 1477/-7.5 1633/-7.5 2025/-12.1
Usage Specifies the frequency and level of the tone.
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Valid Entry 2100/-12.1 2225/-12.1 2804/-16.0
Tone generation: page 2 (TONE GENERATION CUSTOMIZED TONES page) Usage
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Tone Generation
Tone Name
Valid Entry 1-call-wait 2-call-wait 3-call-wait busy busy-verify call-wait-ringback conference confirmation disable-dial hold hold-recall immed-ringback intercept intrusion mntr/rec-warning PBX-dial recall-dial recall-dont-ans redirect reorder rep-confirmation reset-shift ringback secondary-dial special-dial whisper-page zip
Usage Indicates which of the individually administrable tones is modified.
Warning: Customization of "busy-verify" will also change the "warning" tone.
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Chapter 147: Trunk Group
Trunk Group
Sets basic characteristics for every type of trunk group and assigns ports to the group. Many fields are dependent on the settings of other fields and only appear when certain values are entered in other fields on the screen. For example, the entry in Group Type might significantly change the content and appearance of the Trunk Group screen.
Note: For descriptions of the screens and fields that are unique to ISDN trunks, see ISDN Trunk Group.
Trunk Group: page 1
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Trunk Group
Analog Gain
Reduces the strength of incoming signals on TN2199 ports if users regularly experience echo, distortion, or unpleasantly loud volume.
Available only if the country code is 15 and the Trunk Type (in/out) value is 2-wire-ac, 2-wire-dc, or 3-wire.
Valid Entry a b c none
Usage Reduces the incoming signal by -3dB. Reduces the incoming signal by -6dB. Reduces the incoming signal by -8dB. No reduction. This setting should not be changed unless the trunk group sound quality is unacceptable.
Related links Country on page 905 Trunk Type (in/out) on page 918
Auth Code
The Auth Code field is available only for incoming or two-way trunk groups if you enable the Authorization Codes feature. If you enable the Auth Code field, Communication Manager performs an auth code check for the incoming trunk call that is routed over another trunk.
The following table provides the situations when the caller must enter an auth code.
Calling party Station
Called party
Station Trunk
Facility Restriction Level (FRL) check?
No
Yes, if Route Pattern accesses the trunk
Incoming trunk Station
No
Trunk
Yes, if Route Pattern accesses the trunk
Auth Code, required if
No
the FRL of the calling station is less than the FRL of the outgoing route pattern preference
the Auth Code field is enabled on the incoming trunk group
the FRL of the incoming trunk is less than the FRL of the outgoing route pattern preference or the Auth Code field is enabled on the incoming trunk group
Note: The Auth Code field is unavailable if:
· the Group Type field is tandem · the Group Type field is ISDN and the Service Type field is tandem.
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Trunk Group
In these situations, permissions of the caller are transmitted using Traveling Class Mark.
Related links Direction on page 507 Authorization Codes on page 831
BCC
Generalized Route Selection uses the Bearer Capability Class (BCC) to select the appropriate facilities for routing voice and data calls. Far-end tandem servers/switches use the BCC to select outgoing routing facilities with equivalent BCC classes. The BCC entry is used to select the appropriate facilities for incoming ISDN calls. Communication Manager compares the BCC entry with the value of the Bearer Capability information element for the incoming call and routes the call over appropriate facilities.
Available only if all of the following conditions are true:
· ISDN-BRI Trunks or ISDN-PRI are enabled for the system.
· The trunk group type is access, co, fx, tandem, tie, or wats.
· The Comm Type is data, avd, or rbavd.
Valid Entry 0 1 2 4
Usage Voice and voice-grade data 56 kbps synchronous data transmitted with robbed-bit signaling Less than 19.2 kbps synchronous or asynchronous data 64 kbps data on unrestricted channels
Related links Comm Type on page 904 Group Type on page 508 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
Busy Threshold
Valid Entry 1 to 255
Usage
The number of trunks that must be busy to alert attendants to control access to outgoing and two-way trunk groups during periods of high use. When the threshold is reached and the warning lamp for that trunk group lights, the attendant can activate trunk group control: internal callers who dial out using a trunk access code are connected to the attendant. Calls handled by AAR and ARS route patterns go out normally.
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Trunk Group
CDR Reports
Valid Entry y
n r (ring-intvl)
Usage
All outgoing calls on this trunk group generate call detail records. If Record Outgoing Calls Only is disabled for CDR, incoming calls on this trunk group also generate call detail records.
Calls over this trunk group do not generate call detail records.
CDR records are generated for both incoming and outgoing calls. In addition, the following ringing interval CDR records are generated:
· Abandoned calls: The system creates a record with a condition code of "H" indicating the time until the call was abandoned.
· Answered calls: The system creates a record with a condition code of "G" indicating the interval from start of ring to answer.
· Calls to busy stations: The system creates a record with a condition code of " I " indicating a recorded interval of 0.
Related links Record Outgoing Calls Only on page 111
CESID I Digits Sent
For emergency 911 service, Communication Manager might send Caller's Emergency Service Identification (CESID) information to the local telephone company central office or E911 tandem server or switch. This digit string is part of the E911 signaling protocol. Determine the correct entry for this field by talking to the E911 provider. Accepts from one to three digits.
Available only for a cama trunk group.
Related links Group Type on page 508
Comm Type
Indicates whether the trunk group carries voice, data, or both. Note: Comm Types of avd, rbavd, and data require trunk member ports on a DS1 circuit pack.
Valid Entry avd
Usage
For applications that mix voice and Digital Communication Protocol data, such as video conferencing applications. The receiving end server discriminates voice calls from data calls and directs each to an appropriate endpoint. Neither originating nor terminating ends insert a modem pool for any calls. The Signaling Mode for the DS1 circuit pack must be set to either common-chan or CAS signaling.
Table continues...
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Trunk Group
Valid Entry data
rbavd voice
Usage
All calls across the trunk group originate and terminate at Communication Manager digital data endpoints. Public networks don't support data: supported by Avaya's DCP protocol, this entry is used almost exclusively for the data trunk group supporting DCS signaling channels. The Signaling Mode for the DS1 circuit pack must be set to either robbed-bit or common-chan.
For digital trunk groups that carry voice and data with robbed-bit signaling. The Signaling Mode for the DS1 circuit pack must be set to robbed-bit unless mixed mode signaling is allowed on the DS1 circuit pack. In that case, the Signaling Mode must be isdn-ext or isdn-pri.
For trunk groups that carry only voice traffic and voice-grade data (that is, data transmitted by modem). Analog trunk groups must use voice.
Related links Signaling Mode on page 237
COR
Valid Entry 0 to 995
Usage
The Class of Restriction (COR) for the trunk group. Classes of restriction control access to trunk groups, including trunk-to-trunk transfers. Decisions regarding the use of Class of Restriction (COR) and Facility Restriction Levels (FRLs) should be made with an understanding of their implications for allowing or denying calls when AAR/ARS/WCR route patterns are accessed.
Tip:
Remember that FRLs are assigned to classes of restriction. Even if two trunk groups have classes of restriction that allow a connection, different facility restriction levels might prevent operations such as off-net call forwarding or outgoing calls by remote access users.
CO Type
Available only if the country code is 14. Used only by trunk group members administered on a TN464D vintage 2 or later DS1 circuit pack.
Valid Entry analog digital
Usage
Specifies whether the trunk group is connected to analog or digital facilities at the local telephone company central office (CO).
Related links Country on page 905
Country
The country code that corresponds to the protocol used by the local telephone company central office (CO) where the trunk group terminates.
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Trunk Group
Available only for trunk groups that connect Communication Manager to a CO in the public network -- CO, DID, DIOD, FX, and WATS trunk groups.
Valid Entry 1 to 25, except for 19 11
15
18
23
Usage For a list of country codes, see the Country code table.
Communication Manager can be administered for Public Network Call Priority (Call Retention and Re-ring). Communication Manager can be administered for Public Network Call Priority (Intrusion and Re-ring). Communication Manager can be administered for Public Network Call Priority (Mode of Release Control, Forced Disconnect, and Re-ring). If the trunk Group Type is either CO or DID, Communication Manager can be administered for Block Collect Calls.
Related links Country options table on page 817 Trunk Gain on page 935 Trunk Termination on page 935
Dial Access
Controls whether users can route outgoing calls through an outgoing or two-way trunk group by dialing its trunk access code. Allowing dial access does not interfere with the operation of AAR/ ARS.
Security alert:
Calls dialed with a trunk access code over WATS trunks bypass AAR/ARS and are not restricted by facility restriction levels. For security, leave this field disabled unless dial access is needed to test the trunk group.
Valid Entry y n
Usage
Users can gain access to the trunk group by dialing its access code.
Users cannot gain access to the trunk group by dialing its access code. Attendants can still select this trunk group with a Trunk Group Select button.
Digit Absorption List
Valid Entry 0 to 4 blank
Usage
Assigns a digit absorption list, when used, to a trunk group that terminates at a local telephone company central office.
Note:
In a DCS network, DCS features that use the remote-tgs button (on telephones at a remote end) do not work when the incoming trunk group at your end deletes or inserts digits on
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Trunk Group
incoming calls. To manipulate digits in a DCS network (for example, to insert an AAR feature access code), do it on the outgoing side based on the routing pattern.
Direction
The direction of the traffic on this trunk group. Available for all trunk groups except DID and CPE.
Valid Entry incoming outgoing two-way
Usage
Traffic on this trunk group is incoming.
Traffic on this trunk group is outgoing
Traffic on this trunk group is incoming and outgoing. Required for Network Call Redirection.
Related links Disconnect Supervision-Out on page 519 Answer Supervision Timeout on page 517 Receive Answer Supervision on page 649
Group Name
A unique name that provides information about the trunk group. Accepts up to 27 characters.
This field should contain names that identify the vendor and function of the trunk group rather than the group type (DID, WATS).
Note:
Supported by Unicode language display for the 4610SW, 4620SW, 4621SW, and 4622SW, 1600-Series, 9600-Series IP Deskphones, 9600-Series H.323 Deskphones, and Avaya J100 Series IP Phones. Unicode is also an option for the 2420J telephone when the Display Character Set is katakana. For more information about 2420J, see 2420 Digital Telephone User's Guide.
Related links Display Character Set on page 821
Group Number
The trunk group number.
Group Type
The type of trunk group. The fields that are displayed and available might change according to the trunk group type selected.
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Trunk Group Valid Entry Access APLT CAMA CO CPE DID DIOD DMI-BOS FX ISDN
RLT SIP
Tandem
Usage
Used to connect satellite servers to the main switch in Electronic Tandem Networks (ETN). Access trunks do not carry traveling class marks (TCM) and, therefore, allow satellite callers unrestricted access to out-dial trunks on the main server. This entry allows Inband ANI.
Advanced Private Line Termination (APLT) trunks. Used in private networks. This entry allows Inband ANI.
Used to route emergency calls to the local community's Enhanced 911 systems.
Typically used to connect Communication Manager to the local telephone company central office, but can also connect adjuncts such as external paging systems and data modules.
Used to connect adjuncts, such as paging systems and announcement or music sources, to the server running Communication Manager.
Used to direct callers directly to individuals within an organization without going through an attendant or some other central point. This entry allows Inband ANI.
Two-way trunks that are used to transmit dialed digits in both directions. In North America, tie trunks are used for applications that require two-way transmission of dialed digits. This entry allows Inband ANI.
Digital Multiplexed Interface - Bit-Oriented Signaling (DMI-BOS) trunks allow communication with systems using DMI-BOS protocol. This entry also allows Inband ANI.
A local telephone company central office (CO) trunk that connects the server running Communication Manager directly to a CO outside the local exchange area. Used to reduce long-distance charges if the organization averages a high volume of longdistance calls to a specific area code.
Used when digital trunks are needed that can integrate voice, data, and video signals and provide the bandwidth needed for applications such as high-speed data transfer and video conferencing. ISDN trunks can also efficiently combine multiple services on one trunk group.
Also used for Network Call Transfer.
Note:
Available only if ISDN-PRI, ISDN-BRI Trunks, or both have been enabled for the system. For descriptions of the screens and fields that are unique to ISDN trunks, see "ISDN Trunk Group".
Used with Centralized Attendant Service in a private network.
Used to connect a server running Communication Manager to a Session Manager (SM), or to connect two Communication Manager servers.
Note:
The Automatic CallBack, Priority Calling, and Whisper Page features work incorrectly if each of the call's parties is using a SIP endpoint administered on and managed by a different instance of Communication Manager.
Used to connect tandem nodes in a private network. This entry allows Inband ANI.
Table continues...
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Trunk Group
Valid Entry Tie
WATS
Usage
Used to connect a server running Communication Manager to a local telephone company central office or to another server or switch in a private network. Tie trunks transmit dialed digits with both outgoing and incoming calls. This entry also allows Inband ANI.
Used to reduce long-distance bills when your organization regularly places many calls to a specific geographical area in North America. With outgoing WATS service, calls can be sent to certain areas ("WATS band") for a flat monthly charge. Incoming WATS trunks allow toll-free calling to customers and employees.
Related links ISDN-PRI on page 836 ISDN-BRI Trunks on page 835 SBS on page 530 Local Country Code on page 322 International Access Code on page 323 Supplementary Service Protocol on page 521 Carrier Medium on page 505 Path Replacement on page 536 Call Still Held on page 922 Call Still Held on page 644
Incoming Budget
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
This field is available only when you set the ASAC Enabled field on the Multiple Level Precedence & Preemption Parameters screen to y.
Incoming Budget is the number of incoming calls that a trunk group can allow based on the total call limit.
Communication Manager counts the number of active incoming calls and compares the total number of incoming calls with the incoming call budget. If the total number equals or exceeds the budget, Communication Manager replaces established low-precedence calls with new highprecedence calls and blocks new low-precedence calls.
Entire trunk budget (Number of Members) = Incoming budget + Outgoing budget
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Trunk Group
Valid Entry 0255
Blank
Usage
The number of incoming calls that the trunk group can receive. The total of the Incoming Budget and the Outgoing Budget fields must not be greater than the total number of trunk group members.
This field is blank by default.
Incoming call budget high threshold
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
Communication Manager sends an SNMP trap when the system reaches the incoming call budget high threshold.
This field is available only when you set the ASAC Enabled field on the Multiple Level Precedence & Preemption Parameters screen to y.
Valid Entry 0255 Blank
Usage The call limit at which Communication Manager must send an SNMP trap. The field is blank by default.
Incoming call budget low threshold
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
Communication Manager sends an SNMP trap to clear a budget threshold alarm when the system reaches the incoming call budget low threshold.
This field is available only when you set the ASAC Enabled field on the Multiple Level Precedence & Preemption Parameters screen to y.
Valid Entry 0255 Blank
Usage The call limit at which Communication Manager must send an SNMP trap. The field is blank by default.
Incoming Destination
Sets the destination for all incoming calls on trunk groups such as CO, FX, and WATS that must terminate at a single destination. The destination entered here is also the default night service destination. Available only for incoming or two-way trunk groups.
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Trunk Group
Valid Entry Valid extension number
attd blank
Usage
Calls go to this extension number. You can enter any type of extension, though typically the extension entered here identifies a VDN, a voice response unit, or a voice messaging system. Night service overrides this setting when it is active.
Note:
If you enter a shortened extension of the multilocation dial plan, the system does not perform certain administration and validation tasks. Therefore, the system might not display the resultant warnings or submittal denials.
Calls go to the attendant and are recorded as Listed Directory Number (LDN) calls on call detail records.
Leave this field blank if Trunk Type (in/out) is not administered as auto/....
Related links Direction on page 507 Night Service on page 911 Trunk Type on page 936
ITC
Determines the line coding the Generalized Route Selection feature uses for comparison to select appropriate routes for voice and data calls.
Available only when the Comm Type is data, avd, or rbavd and a BCC value is administered.
Valid Entry rest(tricted)
unre(stricted
Usage
Restricted trunks use ami-basic or ami-zcs line coding and can carry only restricted calls.
Unrestricted trunks use b8zs, hdb3, or cmi line coding and can carry restricted or unrestricted calls. A trunk group with an unrestricted ITC can have only unrestricted trunks as members.
Related links Comm Type on page 904
Night Service
Sets the destination for incoming calls when Night Service is operating. If a night service destination is administered for an individual trunk that is a member of this group, that entry overrides the group destination for that trunk. CPE, DID, and DIOD trunk groups do not support night service.
Tip:
Whenever possible, use a night service destination on your switch; otherwise, some features do not function correctly over a DCS network.
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Trunk Group
Valid Entry An extension number (can be a VDN)
attd
blank
Usage The extension of your night service destination.
Note:
If you enter a shortened extension of the multilocation dial plan, the system does not perform certain administration and validation tasks. Therefore, the system might not display the resultant warnings or submittal denials. Calls go to the attendant and are recorded as Listed Directory Number (LDN) calls on call detail records. Leave this field blank if Trunk Type (in/out) is not administered as auto/....
Related links Trunk Type (in/out) on page 918
Number of Members
The system displays this field SIP and H.323 trunk groups.
In Communication Manager Release 8.0 and later, the number of trunk members that can be administered on a SIP trunk group is increased from the current value of 255 to 9999.
Valid entry 0 to 9999
Usage The number of trunks that are members of the Trunk group.
You cannot administer more than 255 trunk members, only if the Number Assignment Method field is set to auto.
Related links Group Type on page 508
Outgoing Budget
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
This field is available only when you set the ASAC Enabled field on the Multiple Level Precedence & Preemption Parameters screen to y.
Outgoing Budget is the number of outgoing calls a trunk group can allow within the total limit.
Communication Manager counts the number of active outgoing calls, and compares the total number of outgoing calls with the outgoing call budget. If the total number equals or exceeds the budget, Communication Manager replaces established low-precedence calls with new highprecedence calls and blocks new low-precedence calls.
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Trunk Group
Valid Entry 0255
Blank
Usage
The number of outgoing calls that the trunk group can make. The total of the Incoming Budget and the Outgoing Budget fields must not be greater than the total number of trunk group members.
The field is blank by default.
Note:
If the total number of Incoming Budget and Outgoing Budget is less than Number of Members, the system displays the following message:
WARNING: Total budget allocated less than total number of trunk group members.
Outgoing call budget high threshold
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
Communication Manager sends an SNMP trap when the system reaches the outgoing call budget high threshold.
This field is available only when you set the ASAC Enabled field on the Multiple Level Precedence & Preemption Parameters screen to y.
Valid Entry 0255 Blank
Usage The call limit at which Communication Manager must send an SNMP trap. The field is blank by default.
Outgoing call budget low threshold
Note:
This field complies with the Unified Capabilities Requirements (UCR) 2008 Change 3 requirements and is approved by Joint Interoperability Test Command (JITC). This field is available only for the USA Department of Defense (DoD) and approved Federal government customers.
Communication Manager sends an SNMP trap to clear a budget threshold alarm when the system reaches the outgoing call budget low threshold.
This field is available only when you set the ASAC Enabled field on the Multiple Level Precedence & Preemption Parameters screen to y.
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Trunk Group
Valid Entry 0255 Blank
Usage The call limit at which Communication Manager must send an SNMP trap. The field is blank by default.
Outgoing Display
With Outgoing Display, telephones can show the name and number of the trunk group used for an outgoing call before the call is connected.
Valid Entry y n
Usage Displays the trunk group name and number. Displays the digits the caller dials.
Prefix-1
If enabled, the prefix "1" is added to the beginning of the digit string for outgoing calls. Use this field for outgoing and two-way trunk groups handling long distance service. Do not enable for trunk groups in AAR or ARS route patterns.
Available only for CO, FX, and DIOD trunk groups.
Protocol Type
The type of line signaling protocol used by the local telephone company central office for DID and DIOD trunk groups.
Available only if the Country code is 15. Used only by trunk group members administered on a TN2199 or TN464D vintage 3 or later circuit pack.
Valid Entry inloc intol
Usage Incoming local. Only the inloc protocol provides ANI. Incoming toll.
Related links Country on page 905 Country options table on page 817
Queue Length
Available only for outgoing or two-way trunk groups.
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Trunk Group
Valid Entry 1 to 100
0
Usage
The number of outgoing calls that can wait in queue when all trunks in a trunk group are busy. Calls wait in queue in the order in which they were made. If a queue is administered, a caller hears a confirmation tone when no trunk is available for the outgoing call. The caller can then disconnect and wait; when a trunk becomes available, Communication Manager calls the extension that placed the original call. Communication Manager remembers the number the caller dialed and automatically completes the call.
Callers receive a busy signal when no trunks are available. Use for DCS trunks.
Related links Direction on page 507
Service Type
The service for which this trunk group is dedicated. In addition to the predefined services or features listed as valid entries, any previously administered user-defined Network Facility Facility Type of 0 (feature) or 1 (service) is allowed.
Valid Entry access accunet cbc
dmi-mos i800 inwats lds megacom
mega800
multiquest
operator
Usage
A tie trunk giving access to an Electronic Tandem Network.
ACCUNET Switched Digital Service -- part of ACI (AT&T Communications ISDN) phase 2.
Call-by-Call service -- provides different dial plans for different services on an ISDN trunk group. Indicates this trunk group is used by the Call-By-Call Service Selection feature.
Digital multiplexed interface -- message-oriented signaling.
International 800 Service -- A subscriber can receive international calls without a charge to the call originating party.
INWATS -- provides OUTWATS-like pricing and service for incoming calls.
Long-Distance Service -- part of ACI (AT&T Communications ISDN) phase 2.
MEGACOM Service -- an AT&T communications service that provides unbanded long-distance services using special access (switch to 4ESS switch) from an AT&T communications node.
MEGACOM 800 Service -- an AT&T communications service that provides unbanded 800 service using special access (4ESS switch to switch) from an AT&T communications node.
AT&T MULTIQUEST Telecommunications Service -- dial 700 service. A terminatinguser's service that supports interactive voice service between callers at switchedaccess locations and service provides directly connected to the AT&T Switched Network (ASN).
Network Operator -- provides access to the network operator.
Table continues...
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Trunk Group
Valid Entry outwats-bnd
public-ntwrk sddn
sdn
sub-operator tandem tie wats-max-bnd
Usage
OUTWATS Band -- WATS is a voice-grade service providing both voice and low speed data transmission capabilities from the user location to defined service areas referred to as bands; the widest band is 5.
Public network calls -- It is the equivalent of CO (outgoing), DID, or DIOD trunk groups. If Service Type is public-ntwrk, Dial Access can be enabled.
Software Defined Data Network -- provides a virtual private line connectivity via the AT&T switched network (4ESS switches). Services include voice, data, and video applications. These services complement the SDN service. Do not use for DCS with Rerouting.
Software Defined Network (SDN) -- an AT&T communications offering that provides a virtual private network using the public switched network. SDN can carry voice and data between customer locations as well as off-net locations.
Presubscribed Common Carrier Operator -- provides access to the presubscribed common carrier operator.
Tandem tie trunks integral to an ET.
Tie trunks -- general purpose.
Maximum Banded Wats -- a WATS-like offering for which a user's calls are billed at the highest WATS band subscribed to by users.
Related links Facility Type on page 630
Signaling Group
Available only for sip trunk groups.
Valid Entry 1 to 999
Usage The number of the SIP signaling group associated with this trunk group.
Related links Group Type on page 508 Group Number on page 708
TAC
The trunk access code (TAC) that must be dialed to access the trunk group. A different TAC must be assigned to each trunk group. CDR reports use the TAC to identify each trunk group. The characters "*" and "# " can be used as the first character in a TAC. Accepts a one- to four-digit number.
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
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Trunk Group
Toll Restricted
If enabled, restricts toll-restricted users from using a trunk access code to make restricted outgoing calls over this trunk group.
Trunk Flash
Enables or disables multifunction telephones to access local telephone company central office (CO) customized services that are provided by servers at the far-end or CO. These CO customized services are electronic features, such as conference and transfer, that are accessed by a sequence of flash signal and dial signals from the Communication Manager telephone on an active trunk call.
Trunk Signaling Type
Controls the signaling used by members in private network trunk groups, mainly in Italy, Brazil, and Hungary. This field also controls the signaling used by members in public network digital trunk groups.
Available only for access, aplt, rlt, tandem, or tie trunk groups.
E&M trunks in Italy, Brazil, and Hungary can use either continuous or discontinuous signaling. Each entry specifies a set of signals and available timers used in the process of setting up and releasing connections. The type of signaling must match the signaling type administered on the far-end server. Use these values only when all trunk group members are assigned to ports on a TN464F, TN2464, or TN2140 circuit pack.
Valid Entry cont dis
Usage Continuous Discontinuous
The following entries are for tie trunks in Main-Satellite/Tributary networks. Use these values only when all trunk group members are assigned to a TN497 circuit pack.
Valid Entry tgu
tge
tgi
Usage
For outgoing trunks, tgu at the main server running Communication Manager administers a tie trunk group connected to a satellite server. (This same group should be administered as tge at the satellite.)
For incoming trunks, tge at a satellite server administers a tie trunk group connected to the main server running Communication Manager. (This same group should be administered as tgu at the main server.)
For internal trunks, tgi administers a two-way tie trunk group between two satellites or between the main server and a satellite. (This trunk group should be administered as tgi on both servers.)
DIOD trunks support pulsed and continuous E&M signaling in Brazil and discontinuous E&M signaling in Hungary. The following entries are for DIOD trunks. Use these values only when all trunk group members are assigned to a TN464F (or later version) or TN2464 circuit pack.
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Trunk Group
Valid Entry cont pulsed discont blank
Usage Continuous E&M signaling. Pulsed E&M signaling. Discontinuous E&M signaling. Leave blank for R2 signaling.
Related links Group Type on page 508
Trunk Type (in/out)
Controls the seizure and start-dial signaling used on this trunk group. Settings might differ for incoming and outgoing trunks.
Valid Entry auto
cont
delay
disc
wink
immed 2-wire-ac 2-wire-dc 3-wire
Usage
Used for immediate connection to a single preset destination (incoming local telephone company central office trunks, for example). No digits are sent, because all calls terminate at the same place.
Continuous signaling is used with Italian E&M tie trunks. The server/switch seizes a trunk by sending a continuous seizure signal for at least the duration specified by the Incoming Seizure Timer.
The sending switch does not send digits until it receives a delay dial signal (an offhook signal followed by an on-hook signal) from the far-end switch, indicating that it is ready to receive the digits.
Discontinuous signaling is used with Italian tie trunks that use E&M signaling. The Avaya server can seize a trunk by sending a single, short signal for the duration specified by the Normal Outgoing Seize Send value. However, with the Three-Way Seizure option, the calling end can also send routing information to the called end by sending one or a series of brief seizure signals.
The sending server or switch does not send digits until it receives a wink start (momentary off-hook) signal from the far-end server or switch, indicating that it is ready to receive the digits.
The sending server or switch sends digits without waiting for a signal from the farend server or switch.
Used with local telephone company central office (CO) trunks in Russia. Select the type of connection to the CO. Check with the network service provider for the type of connection. To use these entries, the Country code must be 15 and the CO trunks must use ports on a TN2199 circuit board.
Tip:
When incoming trunks use the setting immed/immed, the far-end server seizes the trunk and sends digits without waiting for acknowledgment from the receiving end. When traffic is heavy, the receiving server or switch might not immediately attach a Touch Tone Receiver to a call and, therefore, lose digits. Use wink-start trunks or increase the dial-guard timer value on the far-end server or switch to avoid this problem.
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Trunk Group
Related links Country on page 905 Normal Outgoing Seize Send (msec) on page 959
Version
Adjusts the signaling on multi-country local telephone company central office (CO) trunk circuit packs. Entries adjust signaling characteristics on these circuit packs to match the signaling characteristics of the public network in a specific country.
Available only for CO, FX, and WATS trunk groups when the Country code is 5, 16, or 23.
If the Country code is 5, this field controls only TN2147 ports.
Valid Entry a b
Usage
Uses standard signaling for the Netherlands public network.
Uses country 1 (U.S.) signaling. This value is appropriate if Communication Manager is connected to a CO using an Ericcson AXE-10 switch.
If the Country code is 16 or 23, this field sets the input impedance value and only controls TN465C (vintage 2 or later) ports.
Valid Entry a b
Usage Sets input impedance to 600 Ohms. Sets input impedance to 900 Ohms. This value is appropriate in Brazil.
Related links Country on page 905
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Trunk Group
Trunk Group: page 2
Administer Timers
Enables or disables administration of timers on this trunk group. The default for the ISDN trunk group type is disabled. All other trunk group types are enabled by default.
Available for all trunk group types except cpe, h.323, and sip.
Related links Group Type on page 508
Analog Loss Group
Determines which administered two-party row in the loss plan applies to this trunk group if the call is carried over an analog signaling port in the trunk group.
Valid Entry 1 to 17
Usage
The index into the loss plan and tone plan. If values are administered other than in between 6 and 10 or 15 and 17, the system displays a warning message indicating that the loss group may not be appropriate for this trunk group.
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Trunk Group
Answer Supervision Timeout
Valid Entry 0 to 250
Usage
The number of seconds Communication Manager waits before it acts as though answer supervision has been received from the far-end. During a cut-through operation, timing begins after each outgoing digit is sent and timing ceases after the far-end sends answer supervision. On senderized operation, the timer begins after the last digit collected is sent.
Note:
This field's setting does not override answer supervision sent from the network or from DS1 port circuit timers.
Related links Receive Answer Supervision on page 649 Administer Timers on page 516
Auto Guard
Enables or disables Auto Guard, which prevents repeated seizures of a defective trunk. Communication Manager does a maintenance busy-out on these trunks. This field controls ports only on TN438B, TN465B, and TN2147 circuit packs. TN438B ports have hardware support for detecting a defective trunk. TN465B and TN2147 ports use a trunk defective if no dial tone is detected on an outgoing call, and the Outpulse Without Tone feature is disabled for the system.
Available only for co or fx trunk groups.
Related links Group Type on page 508 Outpulse Without Tone on page 326
Bit Rate
Available when the Comm Type is avd or rbavd. Also available if the Comm Type is data, but only if ISDN-PRI is enabled for the system.
Valid Entry 300 1200 2400 4600 9600 19200
Usage The baud rate to be used by pooled modems.
Related links Comm Type on page 904
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Trunk Group
ISDN-PRI on page 836
Caller ID for Service Link Call to H.323 1xC
This field is available only when you set the Direction field to outgoing or two-way and the trunk group type is ISDN or SIP.
Valid entry original-calling-number
station-extension
Usage
Before sending the call to the service link, Communication Manager changes the calling party number to the original calling party number. When the user receives an incoming call, the number that the mobile displays is the original caller ID and not the extension number.
Communication Manager does not change the calling party number.
The default value is station-extension.
Call Still Held
If enabled, the system prevents glare by extending the Incoming Glare Guard timer and delaying an outgoing seizure of a trunk for at least 140 seconds after it is released from an incoming call. This field is used when the receiving end media server or switch initiates the disconnection of incoming calls. This field affects only TN438B, TN465B, and TN2147 ports and is used primarily when the Country code is 2.
Available only for co or fx trunk groups.
Related links Group Type on page 508
Cut-Through
Available only if the Outgoing Dial Type field is rotary or tone. Security alert: Enabling this field reduces the ability to prevent toll fraud.
Valid Entry y
n
Usage
Users get dial tone directly from the far end of the trunk. If you set the Cut-Through field to y, some administered restrictions are bypassed.
Users receive switch dial tone. Instead of digits being sent to the central office, they are collected, and sent all at once when the user finishes dialing.
Related links Outgoing Dial Type on page 930
Cyclical Hunt
Controls the starting point Communication Manager uses to search for an available trunk when a call is sent to the trunk group. Cyclical hunts can be enabled or disabled at any time, however, all the trunks in the group must be idle or busied out.
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Trunk Group
Available only for two-way loop-start trunks.
Valid Entry y
n
Usage
Communication Manager starts its search from the last trunk seized. This method is faster, and therefore suited for high-traffic trunk groups.
Communication Manager starts each search at member 1 (the first trunk administered as a group member).
Related links Direction on page 507 Trunk Type on page 936
Delay Call Setup When Accessed Via IGAR
The system displays this, when the Group Type field is isdn or sip. If a user in one network region accesses an ISDN or SIP trunk in another network region through IGAR, the Delay Call Setup When Accessed Via IGAR field determines whether the IGAR trunk and the outgoing trunk are initiated in sequence or parallel.
Valid Entry y n
Usage The outgoing trunk call waits until the IGAR trunk call is active. The two trunk calls are set up in parallel. The default value is n.
Dial Detection
Indicates whether digit pulses are detected by observing the A-wire (default) or the B-wire only. Applies only to TN2199 ports. The Country code must be 15.
Related links Country on page 905
Digital Loss Group
Valid Entry 1 to 19
Usage
Determines which administered two-party row in the loss plan applies to this trunk group if the call is carried over a digital signaling port in the trunk group. If values other than 18 or between 11 and 15 are administered, the system displays a warning message indicating that the loss group may not be appropriate for this trunk group.
Digits
If the Digit Treatment is absorption, this field specifies how many digits are deleted. If the Digit Treatment is insertion, this field identifies the specific digits that are added.
Valid Entry 1 to 5
Usage The number of digits to be deleted (absorbed).
Table continues...
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Trunk Group
Valid Entry
Up to 4 digits, including * and #
blank
Usage The actual digits to be added (inserted).
This field can be blank only if Digit Treatment is not administered.
Related links Digit Treatment on page 924
Digit Treatment
Modifies an incoming digit string by adding or deleting digits. This is required if the number of digits received does not match the dial plan.
Requires administration of the Digits to add or delete.
Valid Entry absorption insertion blank
Usage Deletes digits, starting at the beginning of the string. Adds digits, starting at the beginning of the string. The incoming digit string is not changed.
Related links Digits on page 923
Disconnect Supervision-In
Indicates whether Communication Manager receives disconnect supervision for incoming calls over this trunk group.
Available only for incoming or two-way trunk groups.
Valid Entry y
n
Usage
Allows trunk-to-trunk transfers involving trunks in this group. The far-end server or switch sends a release signal when the calling party releases an incoming call, and the far-end server or switch is responsible for releasing the trunk. Enhances Network Call Redirection.
The far-end server or switch does not provide a release signal, the hardware cannot recognize a release signal, or timers are preferred for disconnect supervision on incoming calls. Prevents trunk-to-trunk transfers involving trunks in this group.
Caution:
In general, U.S. local telephone company central offices provide disconnect supervision for incoming calls but not for outgoing calls. Public networks in most other countries do not provide disconnect supervision for incoming or outgoing calls. Check with the network services provider.
Related links Trunk Direction on page 649
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Trunk Group
Direction on page 507
Disconnect Supervision-Out
Indicates whether Communication Manager receives disconnect supervision for outgoing calls over this trunk group. Available for outgoing or two-way trunk groups.
Valid Entry y
n
Usage
Allows trunk-to-trunk transfers involving trunks in this group. The far-end sends a release signal when the called party releases an outgoing call, and the far-end is responsible for releasing the trunk. Enhances Network Call Redirection. Available only if Answer Supervision Timeout is 0 and Receive Answer Supervision is enabled.
The far-end server or switch does not provide a release signal, the hardware cannot recognize a release signal, or timers are preferred for disconnect supervision on outgoing calls. Prevents trunk-to-trunk transfers involving trunks in this group.
Caution:
Verify that the far-end server or switch provides answer supervision and disconnect supervision. Most public networks do not provide disconnect supervision over analog trunks. Check with the network services provider.
Related links Direction on page 507 Answer Supervision Timeout on page 517 Receive Answer Supervision on page 649
Disconnect Timing (msec)
Valid Entry
140 to 2550 ms in increments of 10
Usage
Specifies the minimum time in milliseconds that the local telephone company central office or far-end server requires to recognize that this server has disconnected from a call. This timer does not affect ports on a circuit pack that uses the administrable Incoming Disconnect and Outgoing Disconnect timers. Settings on those two timers override this field.
The default of 500 is an industry standard and should not be changed. If this field is set too high, the server or switch does not disconnect sometimes when it should. If this field is set too low, the server or switch disconnects when it should not.
Disconnect Type
Indicates which side or user controls the disconnect. A refers to the calling party, and B refers to the called party.
Available only if the Country code is 15 and the Trunk Type is 2-wire-ac, 2-wire-dc, or 3-wire. Applies only to the TN2199 port.
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Trunk Group
Valid Entry AandB AorB
Usage Both parties control the disconnect. Either party controls the disconnect.
Related links Country on page 905 Trunk Type on page 936
Drop Treatment
Valid Entry intercept busy silence
Usage
Determines what the calling party hears when the called party terminates an incoming call. For security reasons, you must apply a tone; silence can provide an opening for hackers. Applies only to DID trunks.
Note: In Italy, Drop Treatment must be administered as intercept for all DID trunk groups.
Duplex
Available if the Comm Type is avd or rbavd. Also available if the Comm Type is data, but only if ISDN-PRI is enabled for the system.
Note:
Even if the trunk group supports full-duplex transmission, other equipment in a circuit might not.
Valid Entry full half
Usage Allows simultaneous two-way transmission, which is most efficient. Supports only one transmission direction at a time.
Related links Comm Type on page 904 ISDN-PRI on page 836
End-to-End Signaling
Available only for a cpe (customer-provided equipment) trunk group.
Valid Entry
60 to 360 ms in increments of 10
Usage
The duration of the touch tone signal sent to auxiliary equipment. Equipment such as paging equipment and music sources might be connected to Communication Manager by auxiliary trunks. Communication Manager might send DTMF signals (touch tones) to these devices. This field sets the duration of these tones.
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Trunk Group
Related links Group Type on page 508
Expected Digits
Valid Entry 1 to 18
blank
Usage
The number of digits that the far-end server sends for an incoming connection. If the end is absorbing digits on this trunk group, the entry in this field must be larger than the value administered for the number of digits to be absorbed.
Required if Digit Treatment is insert and Digits is a feature access code (for example, AAR or ARS) followed by digits. In this case, the number of digits expected are set on the AAR and ARS Digit Analysis Table and AAR and ARS Digit Conversion Table. If left blank, digit absorption cannot be administered.
Related links Digits on page 923 Digit Treatment on page 924
Extended Loop Range
Available only for a DID trunk group. This field is used only with the TN459A circuit pack.
Valid Entry y
n
Usage
The distance between the local telephone company central office (CO) and the server is greater than the required distance.
The distance between the CO and the server is not greater than the required distance.
Format
Specifies the encoding of Numbering Plan Indicator for identification purposes in the Calling Number and/or Connected Number IEs, and in the QSIG Party Number.
Available only if the Send Calling Number or Send Connected Number field is enabled or restricted.
Valid Entry public unknown private unk-pvt
Usage Indicates that the number plan according to CCITT Recommendation E. 164 is used. Indicates the Numbering Plan Indicator is unknown. Indicates the Numbering Plan Indicator is PNP. Determines the type of number from the private numbering format, but the Numbering Plan Indicator is unknown.
Related links Send Calling Number on page 530 Send Connected Number on page 531
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Trunk Group
Group Type
Displays the type of trunk group.
Related links Group Type on page 508 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
Incoming Calling Number - Delete
Available only for incoming or two-way trunk groups.
Valid Entry 1 to 15 all blank
Usage
The number of digits, if any, to delete from the calling party number for all incoming calls on this trunk group.
Related links Direction on page 507
Incoming Calling Number - Format
The TON/NPI encoding applied to CPN information modified by the CLI Prefix feature. This encoding does not apply to calls originating locally.
If this field is blank, Communication Manager passes on the encoding received in the incoming setup message. If the incoming setup message excludes CPN information and digits are added, the outgoing message contains these digits. If a numbering format is not administered in this case, the value defaults to pub-unk. If the numbering format is administered as unknown, the trunk group is modified to unk-unk encoding of the TON/NPI. Therefore, this field also must contain a value other than unknown.
The values for this field map to the Type of Numbering (TON) and Numbering Plan Identifier (NPI) values shown below.
Valid Entry blank natl-pub intl-pub locl-pub pub-unk lev0-pvt lev1-pvt lev2-pvt unk-unk
Type of Numbering (TON) incoming TON unmodified national(2) international(1) local/subscriber(4) unknown(0) local(4) Regional Level 1(2) Regional Level 2(1) unknown(0)
Numbering Plan Identifier (NPI) incoming NPI unmodified E.164(1) E.164(1) E.164(1) E.164(1) Private Numbering Plan - PNP(9) Private Numbering Plan - PNP(9) Private Numbering Plan - PNP(9) unknown(0)
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Trunk Group
Related links Format on page 927 Format on page 525
Incoming Calling Number - Insert
Valid Entry 0 to 9 all blank
Usage
Up to 15 digits added to the beginning of the digit string of incoming calls when the calling party is a member of this trunk group.
Incoming Dial Tone
Indicates whether or not the server running Communication Manager gives dial tone in response to far-end seizures of the trunk group.
Valid Entry y
n
Usage
Used if the incoming trunk group transmits digits. For example, this option is used for two-way, dial-repeating tie trunks that users select by dialing a trunk access code.
Used for trunks that are not sending digits, such as tandem or incoming local telephone company central office trunks.
Incoming Dial Type
Indicates the type of pulses required on an incoming trunk group. This value should match what the local telephone company central office provides.
Available for Access, APLT, DID, DIOD, DMI-BOS, FX, RLT, Tandem, or WATS trunk groups. Also available for tie trunk groups when the Trunk Signaling Type is blank, cont, or dis.
Valid Entry tone
rotary mf
Usage
Use this entry for Dual Tone Multifrequency (DTMF) addressing, also known as "touchtone" in the U.S. With this entry, trunk group supports both DTMF and rotary signals. Use this entry for the Inband ANI feature. For pulsed and continuous E&M signaling in Brazil and for discontinuous E&M signaling in Hungary, use tone or mf.
Allows only the dial pulse addressing method used by non-touch tone telephones. Though the tone entry supports rotary dialing as well, it is inefficient to reserve touch tone registers for calls that do not use DTMF.
Use this option if the Trunk Signaling Type field is blank. The system displays this field only if you set the Multifrequency Signaling field on the System Parameters Customer-Options screen to y. You cannot use this option if the Used for DCS field is y. For pulsed and continuous E&M signaling in Brazil and for discontinuous E&M signaling in Hungary, use tone or mf.
Table continues...
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Trunk Group
Valid Entry mf2/6
Usage
Use this entry for signaling in Spain. This signaling type provides normal public network service and a virtual private network digital trunk service called IBERCOM. Use this entry if the Trunk Signaling Type field is blank.
The system displays this entry only if you enable Multifrequency Signaling for the system. The system does not display this entry if the trunk is used for DCS.
Related links Group Type on page 508
Incoming Rotary Timeout (sec)
Valid Entry 5 to 99 blank
Usage Sets the maximum time to wait to receive all incoming digits from the far-end switch.
Related links Incoming Dial Type on page 929
Line Length
Available only for trunk groups with a Trunk Signaling Type of tge, tgi, or tgu.
Valid Entry short long
Usage The line length.
Related links Group Type on page 508 Trunk Signaling Type on page 917
Outgoing Dial Type
Sets the method used to transmit digits for an outgoing call. Usually, this method should match what the local telephone company central office provides.
DIOD trunks support pulsed and continuous E&M signaling in Brazil and discontinuous E&M signaling in Hungary.
Available for Access, APLT, CO, DIOD, DMI-BOS, FX, RLT, and WATS trunk groups. Also available for Tie trunk groups when the Trunk Signaling Type is blank, cont, or dis.
Valid Entry tone
Usage
Uses Dual Tone Multifrequency (DTMF) addressing, also known as "touchtone" in the U.S. Allows the trunk group to support both DTMF and rotary signals.
Table continues...
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Trunk Group
Valid Entry rotary automatic mf
mf2/6 r1mf
Usage
Allows only the dial pulse addressing method used by non-touch tone telephones. For example, this value is appropriate for an internal full touch tone system and for a connection to a local telephone company central office that only supports rotary dialing.
For tie trunks if the Trunk Signaling Type is not administered. This provides "cutthrough" operation to outgoing callers who dial a trunk access code, connecting them directly to local telephone company central office dial tone and bypassing any toll restrictions administered on Communication Manager.
Used if a Trunk Signaling Type is not administered.
Available only if Multifrequency Signaling is enabled on the System-Parameters Customer-Options screen. Not available if this trunk is used for DCS.
For pulsed and continuous E&M signaling in Brazil and for discontinuous E&M signaling in Hungary, use tone or mf.
Used if a Trunk Signaling Type is not administered.
Available only if Multifrequency Signaling is enabled on the System-Parameters Customer-Options screen. Not available if this trunk is used for DCS.
Enter r1mf for CAMA trunk groups. It is the only outgoing dial type allowed on CAMA trunk groups. Enter r1mf to allow Russian MF Packet Signaling on outgoing trunks. Russian MF Packet Signaling carries calling party number and dialed number information. Group type field must be set to co.
Related links Group Type on page 508 Multifrequency Signaling on page 837 Trunk Signaling Type on page 917
Preferred Minimum Session Refresh Interval (sec)
The system displays this field only for the SIP trunk groups that do not support Seamless Converged Communications Across Network (SCCAN) calls.
Use this field to set the value of the session refresh timer of non-SCCAN applications. The timer starts after Communication Manager establishes a session with the far-end SIP entities. After every timer interval, Communication Manager sends a session refresh request as a Re-INVITE or an UPDATE message to the SIP entities. If the SIP entities do not receive the request before the interval period expires, the session ends.
Valid entry 90 to 64800
Usage Defines the session refresh timer. The default value is 600.
Note:
If the SIP trunk is a member of a trunk group connected to Session Manager 6.3.6 or earlier, ensure that the value of the field is not more than 1800. If the SIP trunk is not a member, configure Session Manager with an adaptation module for Communication Manager, and set
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Trunk Group
the SessionTimeout parameter on System Manager to a value greater than the value in the Preferred Minimum Session Refresh Interval field.
Related links Group Type on page 508 SCCAN on page 932
Receive Answer Supervision
If enabled, the Communication Manager provides answer supervision for a trunk group. For Outbound Call Management applications, use for trunks supporting network answer supervision. For trunks that do not receive a real answer, this field determines when the CallVisor AdjunctSwitch Application Interface (ASAI) connect event is sent.
Related links Answer Supervision Timeout on page 517 Administer Timers on page 516
Receive Release Ack
If enabled, the Receive Release Ack field is used to set the receive forward and backward release ack options on TN2140 ports (used for Italian and Hungarian tie trunks). Available only if the Trunk Signaling Type is cont or dis.
Related links Trunk Signaling Type on page 917
Redirect on OPTIM failure
Valid Entry
250 to 32000 milliseconds
Usage
Calls are redirected by ROOF (Redirect on OPTIM Failure) procedures if a call terminates to an Extension to Cellular user cell phone (of any application type) that does not have an associated office telephone (or other method of terminating to a physical telephone), or the office telephone is out of order and Communication Manager receives a disconnect prior to the call being answered. The procedure redirects the call so that the caller does not hear ringback forever.
SCCAN
If enabled, this trunk group provides support for incoming SCCAN calls. Available only for sip type trunk groups when Enhanced EC500 is enabled for the system.
Related links Enhanced EC500 on page 833 Group Type on page 508
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Trunk Group
Send Answer Supervision
If enabled, Communication Manager signals the calling server when an incoming call is answered. Available only if the Trunk Signaling Type is cont or dis. The field applies only to TN2140 ports. Available only for incoming or two-way trunks.
Related links Direction on page 507 Trunk Signaling Type on page 917
Send Release Ack
Indicates whether a release acknowledgment is sent in response to a forward or backward release signal. Available only if the Trunk Signaling Type is cont or dis. This field applies only to TN2140 ports (used for Italian and Hungarian tie trunks).
Related links Trunk Signaling Type on page 917
Sig Bit Inversion
Indicates which bits in bit-oriented signaling should be inverted, if any. For trunk ports on TN2242 and TN464B and later circuit packs, this field inverts the A- and B-bits as necessary so that the far-end server or switch can understand seizure and release signals from Communication Manager. If the far-end server, such as a local telephone company central office, on this trunk group interprets the A- and B-bits differently from the default, invert one or both bits -- to change "1" to "0" and vice-versa in the A-bit.
Valid Entry A B A&B none A and none
Usage For the TN464B and later circuit packs.
For the Japanese 2Mbit trunk circuit pack.
Related links Country Protocol on page 225
Supplementary Service Protocol
Available only for ISDN trunk groups.
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Trunk Group
Valid Entry a
b c d e f g
Usage
Allows ASAI Flexible Billing. AT&T, Telcordia Technologies, Nortel.
When the Country code for the DS1 circuit pack is 1A, SSA selects AT&T custom supplementary services.
When the Country code for the DS1 circuit pack is 1B, SSA selects Telcordia Technologies Supplementary Services.
When the Country code for the DS1 circuit pack is 1C, SSA selects Nortel Proprietary Supplementary Services.
QSIG; also used for SBS signaling trunk groups when full QSIG functionality is needed.
ETSI; used for Network Call Deflection.
ECMA QSIG
Allows ASAI Flexible Billing. Allows DCS with rerouting when the trunk is used for DCS and DCS with Rerouting is enabled.
Feature Plus
ANSI. Available only if ISDN-PRI or ISDN-BRI Trunks is enabled for the system, or the trunk is used for DCS. Used for Network Call Transfer.
Related links Country Protocol on page 225 DCS with Rerouting on page 832 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836 Used for DCS on page 953
Synchronization
Available only if the Group Type is :
· dmi-bos or isdn · access, co, fx, tandem, tie, or wats; and the Comm Type is avd or rbavd · access, co, fx, tandem, tie, or wats; the Comm Type is data; and ISDN-PRI or ISDN-BRI
Trunks is enabled for the system
Caution:
Do not change this field without the assistance of Avaya or your network service provider.
Valid Entry async sync
Usage
Determines whether the trunk group uses synchronous or asynchronous communications.
Related links Comm Type on page 904
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Trunk Group
Group Type on page 508 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
Trunk Gain
Specifies the amplification applied to the trunks in this group. With the values administered for Trunk Termination and Country code, the value in this field also determines the input and transhybrid balance impedance for TN465B, TN2146, TN2147, and TN2184 ports. All other CO and DID circuit packs are set automatically to high.
Valid Entry high low
Usage Used if users complain of low volume. Used if users complain of squeal or feedback.
Related links Country on page 905 Trunk Termination on page 935
Trunk Hunt
Defines the trunk hunt search order. Communication Manager performs a trunk hunt when searching for available channels within a facility in an ISDN trunk group. The search can be administered per ISDN-PRI trunk group, but it infers the direction of search within all ISDN-PRI facilities (or portions of those facilities) administered within the trunk group.
Valid Entry ascend cyclical
descend
Usage
Enables a linear trunk hunt search from the lowest to highest numbered channels. All trunks within an ISDN trunk group are selected without regard to the order in which trunks are administered within the trunk group.
Enables a circular trunk hunt based on the sequence the trunks were administered within the trunk group. When using ISDN-BRI interfaces, only cyclical is supported. The cyclical option cannot be set if the trunk group using ISDN-PRI interfaces is to be used for Wideband operations.
Enables a linear trunk hunt search from the highest to lowest numbered channels. All trunks within an ISDN trunk group are selected without regard to the order in which trunks are administered within the trunk group.
Related links Wideband Support on page 534
Trunk Termination
Adjusts the impedance of the trunk group for optimal transmission quality.
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Trunk Group
Valid Entry 600ohm
rc
Usage
The distance to the local telephone company central office (CO) or to the server at the other end of the trunk is less than 3,000 feet.
The distance to the CO or to the server at the other end of the trunk is more than 3,000 feet.
Trunk Type
Controls the seizure and start-dial signaling used on this trunk group. Entries in this field vary according to the function of the trunk group and must match the corresponding setting on the farend server or switch.
Available only for CO, DID, FX, and WATS trunk groups.
Valid Entry ground-start loop-start auto/auto auto/delay auto/immed auto/wink
2-wire-ac 2-wire-dc 3-wire
Usage
Use ground-start signaling for two-way trunks whenever possible. Ground-start signaling avoids glare and provides answer supervision from the far end.
In general, loop-start signaling is used only for one-way trunks. Loop-start signaling is susceptible to glare and does not provide answer supervision.
The term before the slash tells Communication Manager how and when it receives incoming digits. The term after the slash tells Communication Manager how and when it should send outgoing digits.
· auto -- Used for immediate connection to a single preset destination (incoming central office trunks, for example). No digits are sent, because all calls terminate at the same place.
· delay -- The sending server running Communication Manager does not send digits until it receives a delay dial signal (an off-hook signal followed by an on-hook signal) from the far-end server or switch, indicating that it is ready to receive the digits.
· immed -- The sending server running Communication Manager sends digits without waiting for a signal from the far-end server or switch.
· wink -- The sending server running Communication Manager does not send digits until it receives a wink start (momentary off-hook) signal from the far-end server or switch, indicating that it is ready to receive the digits.
These entries are used with local telephone company central office (CO) trunks in Russia. The specific CO should match one of these values.
Available only if the Country code is 15 and the CO trunks use ports on a TN2199 circuit board.
Related links Country on page 905
Unicode Name
Used to determine whether to send Name1 (legacy name) or Name2 (Unicode name). The value for this field is only examined for calls to SIP stations Available only for sip trunk groups.
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Trunk Group
Note:
Avaya BRI deskphones support only ASCII characters because non-ASCII characters, such as Eurofont and Kanafont, show up incorrectly.
Valid Entry y n
auto
Usage Uses the Unicode name. Uses the name as specified on the station form.
Note:
Any non-ASCII characters in the name do not appear correctly on a SIP phone. The choice to use the station name or the Unicode name is automatically determined based on the called phone's capability and the user's preference.
Related links Group Type on page 508
Wink Timer (msec)
Sets the wink timer as follows:
· Sets the maximum duration of the wink signal (wait-for-wink-to-end) when Trunk Type (in/ out) is administered as .../wink.
· Sets the maximum interval after trunk seizure for the wink to begin (wait-for-wink-to-start) when Trunk Type (in/out) is administered as .../delay.
Requirements for the United States domestic network specify that the wink signal for wink-start trunks must begin within 5 seconds after a trunk is seized. For trunks with a delay-dial start, the wink must not last longer than 5 seconds. While some circuit packs are hard-coded to allow the full 5 seconds in both cases, other circuit packs allow you reduce the allowed start time and duration, therefore, reducing the window in which glare can occur.
Available only for wink-type trunks.
Related links Trunk Type (in/out) on page 918
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Trunk Group
XOIP Treatment
Valid Entry auto modem fax tty
Usage
Identifies treatment of an incoming trunk call. When set to modem, fax, or tty, the serving digital signal processor algorithms is placed in immediate pass-through mode. Use this field when the incoming trunk is transporting the XOIP traffic. By identifying this external equipment through administration, VoIP firmware uses this field to determine if it is necessary to immediately attempt to put a call in passthrough mode so that the DSPs do not miss identification of the initial tone events. Default is auto.
Note:
· This feature does not apply to IP trunks (SIP and H.323 trunks).
· This field is intended for exception cases only. For the majority of trunk groups, use the default setting of auto.
Trunk Group: page 3
Caution: Do not change fields on this page without assistance from Avaya or your network service provider.
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Trunk Group
Abandoned Call Search
Indicates whether this trunk group conducts an Abandoned Call Search to identify ghost calls. Abandoned Call Search is designed to work with analog ground-start local telephone company central office (CO) trunks that do not provide disconnect supervision. The CO must support Abandoned Call Search for the feature to work properly. If the CO provides disconnect supervision, the Abandoned Call Search feature is not needed.
Available only for ground-start type trunks.
Related links Trunk Type on page 936
ACA Assignment
Indicates whether Automatic Circuit Assurance (ACA) measurements are taken for this trunk group.
Charge Conversion
Available only for outgoing or two-way CO, DIOD, FX, and WATS trunk groups. Also available for isdn trunk groups, if the Charge Advice field is set to a value other than none.
Valid Entry 1 to 64 500
Usage
Communication Manager multiplies the number of charge units by the value of this field and displays it as a currency amount. Without a value in this field, Communication Manager displays the number of charge units without converting it to currency.
Related links Direction on page 507 Trunk Direction on page 649
Charge Type
Text string used to describe charges related to a telephone call. These words or characters appear on telephone displays after the charge amount. Typically uses either the currency symbol or the charge type, but not both. Accepts up to seven characters. Embedded spaces count as characters.
Available only for outgoing or two-way CO, DIOD, FX, and WATS trunk groups.
Related links Direction on page 507
Connected to CO
Enables or disables overlap sending to a local telephone company central office (CO). Available only for tie trunk groups.
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Trunk Group
Related links Group Type on page 508
Currency Symbol
The telephone displays the symbol before the charge amount. Accepts from one to three characters. Leading and embedded spaces count as characters.
Available only for outgoing or two-way CO, DIOD, FX, and WATS trunk groups. Also available for ISDN trunk groups, if the Charge Advice field is set to a value other than none.
Related links Direction on page 507 Trunk Direction on page 649
Data Restriction
Enables or disables data restriction that is used to prevent tones, such as call-waiting tones, from interrupting data calls. Data restriction provides permanent protection and cannot be changed by the telephone user. Cannot be assigned if Auto Answer is administered as all or acd. If enabled, whisper page to this station is denied.
Decimal Point
The appropriate representation for a decimal point as telephone displays it. Available only with outgoing or two-way CO, DIOD, FX, and WATS trunk groups. Also available for isdn trunk groups, if the Charge Advice field is set to a value other than none.
Note:
If the received charge contains no decimals, no decimal point is displayed (that is, the administered decimal point is ignored for charge information received with no decimals). On a QSIG trunk group, unlike other trunk groups, the Decimal Point field is independent whether the calling display shows a decimal point. Instead, it tells what symbol should be displayed if the QSIG AOC received has a 1/10 or 1/100 or 1/1000 Multiplier.
Valid Entry comma
period
none
Usage
If the received charge contains decimals, the charge is displayed at the calling endpoint's display with a comma as the decimal point. Divides the charge value by 100.
This is the default. If the received charge contains decimals, the charge is displayed at the calling endpoint's display with a period as the decimal point. Divides the charge value by 100.
No decimal point is displayed.
Related links Direction on page 507 Charge Advice on page 506
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Trunk Group
DS1 Echo Cancellation
Enables or disables echo cancellation on a per port basis. If enabled, reduces voice call echo.
Note: Changes to the DS1 Echo Cancellation field is incomplete until one of the following occurs:
· Port is busied-out or released. · Trunk group is busied-out or released. · SAT command test trunk group is performed. · Periodic maintenance runs.
DSN Term
Enables or disables the trunk group as a DSN termination telephone. The default is disabled.
Glare Handling
Determines the reaction of Communication Manager to glare. The following circuit packs can detect glare:
· TN767 (all releases) · TN760C (or later releases) · TN464C (or later releases) Available only for two-way trunks when the outgoing side of the Trunk Type is either .../wink or .../delay.
Valid Entry control
backoff
none
Usage
Communication Manager seizes the trunk and proceeds with call setup. The other switch finds another trunk.
The other server or switch seizes the trunk and proceeds with call setup. The server or switch finds another trunk.
Not administered.
Related links Direction on page 507 Trunk Type on page 936
Hold/Unhold notification
Use this field to enable Communication Manager to send a notification to the far end SIP endpoint when the far end SIP endpoint is put on hold or unhold. When you upgrade to Communication Manager Release 8.1.3, the Hold/Unhold Notifications field is not visible for all SIP trunks and the existing values for all sip trunks are reset to n. For the Hold/Unhold Notifications field to be visible for all SIP trunks, you must enable the SA9143
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Trunk Group
Hold/Unhold Notifications for SIP Trunks field on the system-parameters special-applications screen. . The default value of Hold/Unhold Notifications field on the trunk group for SIP trunks is n. For example, if you add a new SIP trunk when SA9143 Hold/Unhold Notifications for SIP Trunks field is already enabled, the value in the Hold/Unhold Notifications field appears as n. The Hold/Unhold Notifications field appears for all SIP trunks, only if the SA9143 Hold/Unhold Notification for SIP Trunks field on the system-parameters special-applications screen is set to y. Also, if the Hold/Unhold Notifications field for any SIP trunk-group is set to y, then you cannot disable the SA9143 Hold/Unhold Notifications for SIP Trunks field on the system-parameters special-applications screen. For more information, see Avaya Aura® Communication Manager Special Application Features document.
Valid entry y
n
Usage
Communication Manager sends the reINVITE to the far end SIP endpoint to notify the far end SIP endpoint that the endpoint is put on hold or unhold. The media direction in this reINVITE is sendonly when Music On Hold is enabled and is inactive when Music On Hold is disabled. When far end SIP endpoint is put on unhold then reINVITE with media direction sendrecv is send.
Communication Manager does not send the reINVITE to the far end SIP endpoint. The default value is n.
Incoming Tone (DTMF) ANI
Digits received through Automatic Number Identification (ANI) are printed on a CDR record, passed to the Intuity Audix and ASAI interfaces, and displayed on the telephone (and on tandem calls if the outgoing trunk requires ANI). Then the digits are sent to the outgoing trunk.
Available only if the Incoming Dial Type is tone.
Valid Entry *ANI*DNIS*
ANI*DNIS*
no
Usage
If 555-3800 calls extension 81120, the trunk group receives *55538000*81120*. The telephone displays Call from 555-3800.
If 555-3800 calls extension 81120, the trunk group receives 55538000*81120*. The telephone displays Call from 555-3800.
Not administered.
Related links Incoming Dial Type on page 929
Internal Alert
Indicates if internal ringing and coverage is used for incoming calls.
Invoke ID for USNI Calling Name
Communication Manager uses the Invoke ID for USNI Calling Name field to select an Invoke ID each time a calling name is sent over the ISDN trunk group using the USNI protocol.
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Trunk Group
The system displays the Invoke ID for USNI Calling Name field only when:
· The Group Type field is set to ISDN. · The Carrier Medium field is set to PRI/BRI. · The Supplementary Service Protocol field is set to a.
However, the valid entries, variable and fixed-1, have an impact only when the following conditions are met:
· On the Feature-related system parameters screen, the USNI Calling Name for Outgoing Calls field is set to y.
· On the Trunk group screen, the Send Name field is set to y.
Valid entry variable
fixed-1
Usage
Communication Manager selects a new Invoke ID each time a calling name is sent over the ISDN trunk group. This entry is the default value.
Communication Manager selects an Invoke ID with the fixed value 1 each time a calling name is sent over the ISDN trunk group.
Long Holding Time (hours)
Available only if Automatic Circuit Assurance measurements are taken for the trunk group.
Valid Entry 0 to 10
Usage
The number of hours that the system considers a long holding time. A value of 0 indicates that the system does not support long holding calls.
Related links ACA Assignment on page 523
Maintenance Tests
Enables or disables hourly maintenance tests on this trunk group. Available only for aplt, isdn, sip, or tie trunk groups. Related links Group Type on page 508
Measured
Indicates if the system transmits data for this trunk group to the Call Management System.
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Trunk Group
Valid Entry internal
external both
none
Usage Sends the data to the Basic Call Management System (BCMS), the VuStats data display, or both.
Available only if BCMS (Basic) or VuStats is enabled for the system. Sends the data to the CMS. Collects data internally and sends it to the CMS.
Available only if BCMS (Basic) or VuStats is enabled for the system. Trunk group measurement reports are not required.
Related links VuStats on page 847 BCMS (Basic) on page 842
MF Tariff Free
If enabled, Communication Manager generates an MFC Tariff-Free Backward Signal during call setup instead of the "free" signal. This aids local telephone company central office billing.
Available only for Access, APLT, DID, DIOD, DMI-BOS, and Tandem trunk groups when the Incoming Dial Type is mf or for tie trunk groups when the Trunk Signaling Type is blank, cont, or dis, and the Incoming Dial Type is mf.
Related links Group Type on page 508 Incoming Dial Type on page 929 Trunk Signaling Type on page 917
Modify Tandem Calling Number
The system displays this field for the SIP and ISDN trunk groups. This field is available with outgoing or two-way trunks.
Valid Entry natl-intl-prefix
tandem-cpn-form
no
Usage
Adds the national or international prefixes from the Locations Parameters screen when the calling party number is appropriate.
Modifies the calling party number IE in the previously administered format specified for the Tandem Calling Party Number.
Does not modify the calling party number.
Related links Modify Tandem Calling Number on page 527
Numbering Format
Available only for sip trunk groups.
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Trunk Group
Valid Entry public private
Usage Indicates that the number plan according to CCITT Recommendation E. 164 is used. Indicates the Numbering Plan Indicator is PNP.
Outgoing ANI
The digit string sent in place of normal ANI. Overrides the normal ANI if this trunk group is used for an outgoing call with ANI. The ANI is sent exactly as administered, except for the normal truncation to seven digits for Russian ANI. This ANI override works both for calls originated in Communication Manager and calls tandemed through it. Accepts up to 15 digits.
Available only for CO, DIOD, FX, and WATS trunk groups.
Path Replacement Method
Available only if the Group Type is ISDN, the Supplementary Service Protocol is b or e, and Supplementary Services with Rerouting or DCS with Rerouting is enabled for the system. Unavailable if Path Replacement with Retention is enabled.
Valid Entry always
BR (better route)
Usage
Use any QSIG (SSB) trunk group as the replacement trunk group. A new call is always originated, even when the original trunk group is determined to be the replacement trunk group.
Route pattern preferences help determine trunk group path replacement. The original trunk group is retained if Path Replacement with Retention is enabled. Path replacement fails (and the original trunk group is retained) if Path Replacement with Retention is disabled.
Related links Group Type on page 508 Supplementary Service Protocol on page 521 Supplementary Services with Rerouting on page 849 DCS with Rerouting on page 832 Path Replacement with Retention on page 537
Path Replacement with Retention
Available only if the Group Type is ISDN, the Supplementary Service Protocol is b or e, and Supplementary Services with Rerouting or DCS with Rerouting is enabled for the system.
Valid Entry y n
Usage
Retains the original trunk group.
Provides path replacement according to settings for the Path Replacement Method.
Related links Supplementary Service Protocol on page 521
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Trunk Group
Supplementary Services with Rerouting on page 849 DCS with Rerouting on page 832 Group Type on page 508
PBX ID
Valid Entry 1 to 63 blank
Usage
Identifies the remote switch in the network with which the trunk communicates on a DCS signaling link. Available only for trunks that are used for DCS.
Related links Used for DCS on page 953
Per Call CPN Blocking Code
A one- to four-digit number used for access, APLT, CO, DIOD, FX, tandem, tie, and WATS trunk groups only. The characters * and # can be used as the first digit.
Per Call CPN Unblocking Code
A one- to four-digit number used for access, APLT, CO, DIOD, FX, tandem, tie, and WATS trunk groups only. The characters * and # can be used as the first digit.
Precedence Incoming
Available only when the trunk group is a DSN termination telephone and the trunk group type is tie.
Valid Entry digit
dtmf (a-d)
Usage
Precedence level for dual-tone multifrequency (DTMF) or tone trunks is received as digits (rotary pulses).
Precedence level for dual-tone multifrequency (DTMF) or tone trunks is received as DTMF signals (touchtones).
Related links DSN Term on page 525 Group Type on page 508
Precedence Outgoing
Available only when the trunk group is a DSN termination telephone and the trunk group type is tie.
Valid Entry digit
dtmf (a-d)
Usage
Precedence level for dual-tone multifrequency (DTMF) or tone trunks is sent as digits (rotary pulses).
Precedence level for dual-tone multifrequency (DTMF) or tone trunks is sent as DTMF signals (touchtones).
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Trunk Group
R2 MFC Signaling
Available only if: · Multinational Locations is enabled for the system, and either of the following: - Outgoing Dial Type is mf. - Incoming Dial Type or Outgoing Dial Type is rotary, and Country code is 15 (Russia).
Valid Entry 1 to 8
Usage The MFC signaling parameters set used by this trunk group.
Related links Country on page 905 Multinational Locations on page 838 Incoming Dial Type on page 929 Outgoing Dial Type on page 930
Receive Analog Incoming Call ID
Enables or disables the collection of incoming call ID information on analog trunks. Fifteen characters of name and number information associated with an incoming call (ICLID, or incoming call line identification information) is stored and displays.
Available for CO, DID, and DIOD trunk groups when Analog Trunk Incoming Call ID is enabled. The trunk must be incoming or two-way.
Valid Entry Bellcore NTT disabled V23Bell
Usage Collects ICLID information in the U.S. Collects ICLID information in Japan. Stops the collection of ICLID information on analog trunks. For Telcordia Technologies protocol with V.23 modem tones. Used in Bahrain and similar countries.
Related links Analog Trunk Incoming Call ID on page 828 Direction on page 507
Replace Unavailable Numbers
If enabled, replaces unavailable numbers with administrable strings for incoming and outgoing calls assigned to the specified trunk group. Applies to BRI/PRI, H.323, and SIP trunks. Also applies to analog trunks if Analog Trunk Incoming Call ID is enabled for the system and Receive Analog Incoming Call ID for the trunk is set to any value except disabled.
Available only if the group type is isdn or sip.
Related links Analog Trunk Incoming Call ID on page 828
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Trunk Group
Group Type on page 508 Receive Analog Incoming Call ID on page 947
Request Category
Indicates if Communication Manager should request a call category from the local telephone company central office (CO).
Available only if the Country code is 15 and MF shuttle signaling is enabled.
Related links Country on page 905 Shuttle on page 952
Seize When Maintenance Busy
Indicates whether this server generates an outgoing seizure when a trunk in this trunk group is maintenance busied and whether the far-end server or switch is administered to do likewise. This supports the Electronic Tandem Network Busyout feature, which is intended to prevent a far-end server or switch from reporting problems with a trunk that has been removed from service. This field does not affect the behavior of the far-end server or switch. It controls the behavior of your server and defines the expected far-end behavior.
This field only affects ports on TN760C (or later release), TN767, and TN464C (or later release) circuit packs. For DIOD trunks using TN464F (or later release) or TN2464 circuit packs, available only for diod trunk groups when the Trunk Signaling Type is pulsed, cont, or dis.
Valid Entry near-end
far-end both-ends
Usage
Communication Manager generates an outgoing seizure when a trunk is maintenance busied, but the far-end server or switch does not. The seizure is maintained until the maintenance busyout is released.
The far-end server or switch generates an outgoing seizure when a trunk is maintenance busied, but this server running Communication Manager does not.
Both this server running Communication Manager and the far-end server or switch generate an outgoing seizure when a trunk is maintenance busied.
If a server generates an outgoing seizure when a trunk is busied out, the seizure will probably cause alarms at the far-end server or switch, perhaps leading to a far-end maintenance busy out, unless the far-end server or switch is administered to expect this behavior.
If the administered value of this field is either far-end or both-ends, any abnormally long incoming seizure (including failure to drop from a completed call) is the result of a far-end maintenance busy condition. This assumption might be incorrect, since the abnormally long seizure might actually be due to failure of the trunk circuit.
Related links Group Type on page 508 Trunk Signaling Type on page 917
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Trunk Group
Send Called/Busy/Connected Number
Specifies if the dialed number, whether called (ringing), busy (busy tone), or connected (answered) is sent on incoming or tandemed ISDN calls.
Available only if QSIG Value-Added is enabled for the trunk group.
Valid Entry y
n r
Usage
The dialed number is sent on incoming or tandemed ISDN calls. This field must be enabled in order for the Calling Party Number of an incoming ISDN call to display at the transferred-to station after a QSIG transfer operation. If enabled, the Numbering - Public/Unknown Format is accessed to construct the actual number sent, or the Numbering - Private Format is used.
Disables the sending of the dialed number on incoming or tandemed ISDN calls.
Restricted. The connected number is sent "presentation restricted".
Related links Numbering -- Public/Unknown Format on page 636 QSIG Value-Added on page 537 Numbering-Private Format on page 633
Send Calling Number
Specifies whether the calling party's number is sent on outgoing or tandemed ISDN calls.
Note: The Numbering - Public/Unknown Format can override the Send Calling Number administration
Valid Entry y
n
r
Usage
The calling party's number is sent on outgoing or tandemed ISDN calls. If enabled, the Numbering - Public/Unknown Format is accessed to construct the actual number sent, or the Numbering - Private Format is used.
Disables the sending of the calling party's number on outgoing or tandemed ISDN calls. If disabled, an incoming number is not tandemed out again. This applies to all Supplementary Service Protocols.
Restricted. The calling number is sent "presentation restricted". If set to restricted, an incoming number is marked restricted when it is tandemed out again. This applies to all Supplementary Service Protocols.
Related links Numbering-Private Format on page 633 Incoming Number Format on page 100 Numbering -- Public/Unknown Format on page 636
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Trunk Group
Send Connected Number
Specifies if the connected party's number is sent on incoming or tandemed ISDN calls. Available only if QSIG Value-Added is disabled for the trunk group.
Valid Entry y
n r
Usage
The connected party's number is sent on outgoing or tandemed ISDN calls. If enabled, the Numbering - Public/Unknown Format is accessed to construct the actual number sent, or the Numbering - Private Format is used. This field must be enabled for the Calling Party Number of an incoming ISDN call to display at the transferred-to station after a QSIG transfer operation.
Disables the sending of the connected party's number on outgoing or tandemed ISDN calls. If disabled, an incoming number is not tandemed out again. This applies to all Supplementary Service Protocols.
Restricted. The connected number is sent "presentation restricted". If this field is set to r, an incoming number is marked restricted when it is tandemed out again. This applies to all Supplementary Service Protocols.
Note:
The AT&T Switched Network Protocol does not support restricted displays of connected numbers. Therefore, if you administer the 1a country-protocol/ protocol-version combination for the DS1 Circuit Pack, you should not administer the Send Connected Number as restricted, as this causes display problems. The Numbering - Public/Unknown Format overrides the Send Connected Number administration for any administrable block of extensions.
Related links Numbering-Private Format on page 633 QSIG Value-Added on page 537 Numbering -- Public/Unknown Format on page 636
Send EMU Visitor CPN
Controls which calling party identification (extension of the primary telephone or extension of the visited telephone) is used when a call is made from a visited telephone. There are areas where public network trunks disallow a call if the calling party information is invalid. In this case, there can be instances where the extension of the primary telephone is invalid and the extension of the visited telephone must be used.
Valid Entry y
n
Usage
Sends calling party identification information on the extension of the EMU user's telephone.
Sends calling party identification information on the primary telephone.
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Trunk Group
Send Name
Specifies whether the calling, connected, called, or busy party's administered name is sent to the network on outgoing or incoming calls. Available only for isdn or sip trunk groups.
Valid Entry y
n
r (restricted)
Usage
When the Supplementary Service Protocol is e (DCS with Rerouting), only values of y and n are permitted.
When the Supplementary Service Protocol is e (DCS with Rerouting), only values of y and n are permitted.
The calling/connected name will be sent by Communication Manager, but will be marked "presentation restricted". This value is valid only if the Supplementary Service Protocol is a (national supplementary service), b (for called/busy only), or d for the QSIG Global Networking Supplementary Service Protocol.
Note:
If name information is not administered for the calling station or the connected, called, or busy station; the system sends the extension number instead of the name.
Related links Group Type on page 508 Supplementary Service Protocol on page 933
Short Holding Threshold
Available only if Automatic Circuit Assurance measurements are taken for this trunk group.
Valid Entry 0 to 30
Usage The number of times the system will record a short holding call before alerting an attendant to the possibility of a faulty trunk.
If 0 is entered, no short holding calls are recorded.
Related links ACA Assignment on page 523
Short Holding Time (seconds)
Available only if Automatic Circuit Assurance measurements are taken for this trunk group.
Valid Entry 0 to 160
Usage The length of time that the system considers as being a short holding time. If 0 is entered, the system does not support short holding calls.
Related links ACA Assignment on page 523
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Trunk Group
Show ANSWERED BY on Display
Available only for isdn pri/bri, isdn h.323, and sip trunk groups.
Valid Entry y
n
Usage
The words "ANSWERED BY" display in addition to the connected telephone number on calls over this trunk. This is the default.
Note:
Based on display language settings for stations, "ANSWERED BY" is translated into and displayed in the appropriate language.
Only the connected telephone number displays. This might be preferred when outgoing calls are over a trunk that might be redirected.
Shuttle
Enables or disables MF shuttle signaling. It can be administered on TN464D (or later release) or TN2199 circuit packs.
Available only for co, fx, or wats trunk groups, with a Country code of 15, when the Outgoing Dial Type is rotary.
Related links Country on page 905 Group Type on page 508 Outgoing Dial Type on page 930
Signaling Group
The signaling group number.
Start B Signal
Indicates which B-signal should be used to start a call. The value administered in this field must be coordinated with the local telephone company central office. Available only when the Country code is 15 and MF shuttle signaling is enabled.
Valid Entry 1 2 3
Usage Start calls with signal B1 (first digit). Start calls with signal B2 (next digit). Start calls with signal B3 (previous digit).
Related links Country on page 905 Shuttle on page 952
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Trunk Group
Start Position
Available only when the Country code is 15 and MF shuttle signaling is enabled.
Valid Entry 1 to 9
Usage
Indicates which digit in the digit string is the previously sent digit. The value administered in this field must be coordinated with the local telephone company central office.
Related links Country on page 905 Shuttle on page 952
Suppress # Outpulsing
Indicates whether or not to suppress the final "#" in cases where the system would normally outpulse it. Used if end-to-end signaling begins with (and includes) "#". This field should be enabled when the local telephone company central office or any other facility treats "#" as an error.
Time (sec) to Drop Call on No Answer
Available only for co or diod trunk groups when the Outgoing Dial Type is mf, or for co, diod, fx, or wats trunk groups when the Country code is 15.
Valid Entry 0 to 1200
Usage
The duration in seconds that Communication Manager should wait for outgoing calls to be answered. If the call is unanswered in the specified number of seconds, the call drops.
If 0 is entered, the timer is not set and no calls drop.
Related links Country on page 905 Group Type on page 508 Outgoing Dial Type on page 930
Used for DCS
If enabled, this trunk group sends and receive messages on a DCS signaling link.
Note:
This field cannot be enabled if the trunk group number is greater than 255 or if the Trunk Access code is more than 3 digits long.
Related links TSC Supplementary Service Protocol on page 721 DCS with Rerouting on page 832 ISDN-BRI Trunks on page 835 ISDN-PRI on page 836
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Trunk Group
Group Type on page 708
Used Only for Paging
Indicates whether or not this trunk is used only for paging. Default is disabled.
Available only for wats trunk groups when Port Network Support is disabled for the system.
Related links Group Type on page 508 Port Network Support on page 839
Voice Paging Timeout (sec)
Available only if the trunk is used only for paging.
Valid Entry 10 to 6000
Usage The number of seconds before a paged trunk call drops. Default is 10.
Related links Used Only for Paging on page 954
Wideband Support
Note:
This feature is not supported on the DS1 interfaces on branch gateways (G700).
Enables or disables wideband switching on this trunk group. Only trunk members from TN464C or later circuit packs can use wideband switching.
Available only if Wideband Switching is enabled for the system.
Note:
Wideband trunk calls are treated as a single trunk call when Automatic Circuit Assurance (ACA) measurements are taken. This way, if an ACA referral call is generated (for short or long holding time), the wideband call only triggers a single referral call using the lowest Bchannel trunk member associated with the wideband channel.
Related links Wideband Switching on page 842
Administrable Timers
The system might not display this screen for all trunk group types.
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Administrable Timers
Caution: Customers: Do not change fields on this page without assistance from Avaya or your network service provider.
Answer Send (msec)
Available only for incoming or two-way trunk groups when the Trunk Type is dis. Only TN2140 and TN2199 ports receive this timer.
Valid Entry
10 to 2550 in increments of 10
Usage The duration of the answer signal pulse.
Related links Direction on page 507 Trunk Type on page 936
Busy Tone Disconnect
If enabled, Communication Manager recognizes a busy tone signal as a disconnect on this trunk group.
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Trunk Group
Available only if Enable Busy Tone Disconnect for Analog loop-start Trunks is enabled for the system country options. Related links Enable Busy Tone Disconnect for Analog Loop-start Trunks on page 821
Cama Outgoing Dial Guard (msec)
Available only for cama trunk groups (the trunk group type used for emergency 911 service).
Valid Entry
25 to 6375 in increments of 25
Usage
Minimum interval between the seizure acknowledgment on the receiving server or switch and the outpulsing of digits by this server.
Related links Group Type on page 508
Cama Wink Start Time (msec)
Available only for cama trunk groups.
Valid Entry
20 to 5100 in increments of 20
Usage
The duration (the wait-for-wink-to-end time) for a wink-start CAMA trunk. The wink must begin before the Outgoing Seizure Response timer expires.
Related links Group Type on page 508
Disconnect Signal Error (sec)
The system displays this field only for ground-start trunk groups.
Valid entry
1 to 255 in increments of 1
Usage
The maximum interval that Communication Manager waits to receive a disconnect signal from the far-end after the local party, such as a telephone or tie trunk, goes on-hook.
If the timer expires, the disconnect signal fails.
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Administrable Timers
Flash Length (msec)
Valid Entry
10 to 2550 in increments of 10
Usage
The duration of a flash signal generated toward the local telephone company central office. This timer is sent to TN436B, TN459B, TN464C (or later), TN465B (or later), TN753 (if Country code is 23), TN2146, TN2147, TN2184, and TN2199 circuit boards.
Glare
Available only for two-way or outgoing trunk groups when the Trunk Type is cont. Only TN2140 ports receive this timer.
Valid Entry
40 to 100 in increments of 10
Usage
The minimum acceptable interval between the moment your server running Communication Manager sends an outgoing seizure and the moment it receives a seizure acknowledgment. If acknowledgment is received before the timer expires, glare is assumed.
Related links Direction on page 507 Trunk Type on page 936
Incoming Dial Guard (msec)
Valid Entry
10 to 2550 in increments of 10
Usage
The minimum acceptable interval between the detection of an incoming seizure and the acceptance of the first digit. Communication Manager does not accept digits before this timer expires. This timer is never sent to TN429 ports.
Incoming Disconnect (msec)
Available only for incoming or two-way trunk groups when the Trunk Type is blank or cont.
Valid Entry
50 to 2550 in increments of 10
Usage
The minimum valid duration of a disconnect signal for an incoming call. Communication Manager does not recognize shorter disconnect signals. This field cannot be blank. For Brazil pulsed E&M signaling, use 600.
Related links Direction on page 507
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Trunk Group Trunk Type on page 936
Incoming Disconnect Send (msec)
Available only for incoming or two-way trunk groups when the Trunk Type is dis. Only TN2140 ports receive this timer.
Valid Entry
500 to 1200 in increments of 100
Usage
The duration of the backward release signal the server running Communication Manager sends at the end of an incoming call.
Related links Direction on page 507 Trunk Type on page 936
Incoming Glare Guard (msec)
Available only for two-way trunk groups.
Valid Entry
100 to 25500 in increments of 100
Usage
The minimum interval that must elapse between a trunk's release from an incoming call and its seizure for an outgoing call. This delay gives the far-end time to release all equipment after the trunk is released. This field cannot be blank.
Related links Direction on page 507
Incoming Incomplete Dial Alarm (sec)
Valid Entry
1 to 255 in increments of 1
Usage
The maximum acceptable interval between an incoming seizure and receipt of all digits. Intervals greater than this limit generate an inline error. Only the TN436 (all), TN459 (all), TN464C (or later), TN767, TN2140, TN2146, TN2184, TN2199, and TN2242 circuit packs use this timer.
Incoming Partial Dial (sec)
Available only if the Incoming Dial Type is rotary. This timer is never sent to TN429 ports.
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Administrable Timers
Valid Entry
5 to 255 in increments of 1
Usage The maximum time allowed between incoming rotary digits.
Related links Incoming Dial Type on page 929
Incoming Seizure (msec)
Available only for incoming or two-way trunk groups, and, when applicable, the Trunk Type is cont. Only TN429, TN438 (any release), TN 447, TN464C (or later), TN465 (any release), TN767, TN2138, TN2140, TN2147, TN2184, and TN2199 ports receive this timer. For DID trunks, only TN2199 and TN429D (or later) receive this timer.
Valid Entry
20 to 2550 in increments of 10
Usage
The duration of the shortest incoming seizure signal the server running Communication Manager can recognize. For ICLID, set this field to 120. The field cannot be blank.
Related links Direction on page 507 Trunk Type on page 936
Normal Outgoing Seize Send (msec)
Available only for two-way or outgoing trunk groups when the Trunk Type is dis. Only TN2140 ports receive this timer.
Valid Entry
10 to 990 in increments of 10
Usage
The duration of the signal the server running Communication Manager sends for an outgoing seizure.
Related links Direction on page 507 Trunk Type on page 936
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Trunk Group
Outgoing Dial Guard (msec)
Valid Entry
100 to 25500 in increments of 100
Usage
The minimum interval between seizure acknowledgment of a trunk and the outpulsing of digits. For trunks that do not provide seizure acknowledgment, the timer specifies the minimum time between seizure and the outpulsing of digits. Any digit the caller dials after lifting the receiver, but before the timer expires, is not outpulsed until the timer expires.
This field cannot be blank.
Outgoing Disconnect (msec)
Valid Entry
50 to 2550 in increments of 10
Usage
The minimum valid duration of a disconnect signal for an outgoing call. Communication Manager does not recognize shorter disconnect signals. This timer begins timing when a disconnect signal is detected on an outgoing call and resets when the signal is no longer detected. If the timer expires, the trunk drops.
This field cannot be blank. For Brazil pulsed E&M signaling, use 600.
Outgoing Disconnect Send (msec)
Available only for two-way or outgoing trunk groups when the Trunk Type is dis. Only TN2140 ports receive this timer.
Valid Entry
100 to 9900 in increments of 100
Usage
The duration of the forward release signal the server running Communication Manager sends at the end of outgoing calls.
Related links Direction on page 507 Trunk Type on page 936
Outgoing End of Dial (sec)
Available for outgoing or two-way trunk groups when the network does not provide answer supervision.
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Administrable Timers
Valid Entry
1 to 254 in increments of 1
Usage
The maximum time Communication Manager waits to receive answer supervision for outgoing calls on the ports controlled by firmware timers. Controls firmware answer supervision timers on circuit packs that have them.
During a cut-through operation, timing begins after Communication Manager sends each outgoing digit and ceases when answer supervision is received. If the timer expires, Communication Manager acts as if it has received answer supervision. On senderized operation, the timer begins after the switch sends the last digit collected. The timer ceases when answer supervision is received. If the timer expires, Communication Manager acts as if it has received answer supervision.
For Brazil pulsed E&M signaling, use 40.
Related links Direction on page 507 Receive Answer Supervision on page 649
Outgoing Glare Guard (msec)
Available only for outgoing and two-way trunk groups.
Valid Entry
100 to 25500 in increments of 100
Usage
The minimum interval that must elapse between a trunk's release from an outgoing call and its seizure for another outgoing call. This delay gives the far-end time to release all equipment after the outgoing trunk is released. This field cannot be blank.
Related links Direction on page 507
Outgoing Last Digit (sec)
Available only for two-way or outgoing trunk groups when the Trunk Type is dis or cont. Only TN497 and TN2140 ports receive this timer.
Valid Entry
1 to 40 in increments of 1
Usage
The maximum time that Communication Manager waits for the next digit dialed. After the timer expires, no more digits are accepted by the circuit pack.
Related links Direction on page 507 Trunk Type on page 936
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Trunk Group
Outgoing Rotary Dial Interdigit (msec)
Available only if:
· The trunk Group Type is access, aplt, co, diod, dmi-bos, fx, rlt, tandem, or wats, and the Outgoing Dial Type is rotary.
· The trunk Group Type is tie, the Trunk Type is blank, cont, or dis, and the Outgoing Dial Type is rotary.
· The trunk Group Type is tie, and the Trunk Type is tge, tgi, or tru.
Valid Entry
150 to 2550 in increments of 10
Usage The minimum time between outpulsed digits on outgoing rotary trunks.
Related links Group Type on page 508 Outgoing Dial Type on page 930 Trunk Type on page 936
Outgoing Seizure (msec)
Available only if the Country code is 15, for outgoing or two-way trunk groups, when the Trunk Type is 2-wire-ac, 2-wire-dc, or 3-wire. This timer is sent only to the TN2199 circuit pack.
Valid Entry
20 to 2550 in increments of 10
Usage The duration of the outgoing seizure signal.
Related links Country on page 905 Direction on page 507 Trunk Type on page 936
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Administrable Timers
Outgoing Seizure Response (sec)
Valid Entry
1 to 255 in increments of 1
Usage
The maximum interval that Communication Manager waits after sending a seizure signal to receive seizure acknowledgment from the far-end. If the acknowledgment is not received within this time, a seizure failure response is uplinked. This timer is sent to the TN438B, TN439, TN447, TN458, TN464B (or later), TN465B (or later), TN767, TN2140, TN2147, TN2184, TN2199, and TN2242 circuit packs.
For Brazil pulsed E&M signaling, use 255.
Programmed Dial Pause (msec)
Valid Entry
100 to 25500 in increments of 100
Usage
The exact duration of the pause used during abbreviated dialing, ARS outpulsing, and terminal dialing operations. This timer is administrable for all outgoing and twoway trunk groups. This timer works with the TN464B (or later), TN767, TN458, TN2140, and TN2242 tie circuit packs. All central office (CO) circuit packs that accept administrable timers accept this timer.
Release Ack Send (msec)
Available only for incoming and two-way trunk groups when the Trunk Type is dis. Only TN2140 ports receive this timer.
Valid Entry
500 to 1200 in increments of 100
Usage
The duration of the signal the server running Communication Manager sends for a forward release acknowledgment. After the server running Communication Manager receives a forward release signal, it must send a forward release acknowledgment signal.
Related links Direction on page 507 Trunk Type on page 936
Ringing Monitor (msec)
The minimum time that Communication Manager requires to determine if a trunk disconnects. You cannot leave the field blank.
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Trunk Group
Valid entry
200 to 51000 in increments of 200
Usage
If the ringing signal disappears for a duration longer than the time specified in this field, you can be sure that the call is disconnected. This timer is sent to TN464C (or later), TN767, TN438 (all), TN447, TN465 (all), TN2138, TN2147, TN2184, and TN2199 CO circuit packs.
Seize Ack Delay (msec)
Available only for incoming or two-way trunk groups if the Trunk Type is dis. Only TN2140 ports receive this timer.
Valid Entry
40 to 120 in increments of 10
Usage
The maximum interval the server running Communication Manager waits after receipt of an incoming seizure to send seizure acknowledgment.
Related links Direction on page 507 Trunk Type on page 936
Seize Ack Send (msec)
Available only for incoming or two-way trunk groups if the Trunk Type is dis. Only TN2140 ports receive this timer.
Valid Entry
10 to 990 in increments of 10
Usage
The duration of the seizure acknowledgment signal the server running Communication Manager sends in response to an incoming seizure.
Related links Direction on page 507 Trunk Type on page 936
Send Incoming/Outgoing Disconnect Timers to TN465 Ports
If enabled, the incoming disconnect and outgoing disconnect timer values are sent to the trunk group ports that are on a TN465 board. Available only for a co, fx, or wats trunk group.
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Administrable Timers
END TO END SIGNALING
Pause (msec)
Available only if the Trunk Type is blank.
Valid Entry
20 to 2550 in increments of 10
Usage
The interval (pause) between DTMF tones sent from a hybrid telephone. All CO, DIOD, and tie circuit packs that accept administrable timers accept this timer. However, this timer is sent only to the following circuit packs: TN464B (or later), TN767, TN436B, TN459B, TN2146, TN2199, and TN2242, and TN429 and TN2184 ports in a DID trunk group.
Related links Trunk Type on page 936
Tone (msec)
Available only if the Trunk Type is blank.
Valid Entry
20 to 2550 in increments of 10
Usage
The duration of the DTMF tone sent when a button on a hybrid telephone is pressed. All CO, DIOD, and Tie circuit packs that accept administrable timers accept this timer. This timer is also sent to the following circuit packs: TN464B (or later), TN767, TN436B, TN459B, TN2146, TN2199, TN429, TN2184 ports in a DID trunk group.
Related links Trunk Type on page 936
OUTPULSING INFORMATION
Break (msec)
The duration of the break interval (the pulse duration) while the system is outpulsing digits using dial pulse signaling. The field cannot be blank.
Valid Entry
20 to 80 in increments of 5
10 to 40 in increments of 5
Usage
If the PPS (pulses per second) value is 10, the sum of the Make (msec) and Break (msec) values must equal 100.
If the PPS (pulses per second) value is 20, the sum of the Make (msec) and Break (msec) values must equal 50.
Related links Make (msec) on page 966 PPS on page 967
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Trunk Group
Frequency
Identifies the Periodical Pulse Metering (PPM) pulse frequency, or frequencies, sent by the public network. Circuit packs can detect up to three different frequencies (12kHz, 16kHz, and 50Hz), plus two frequency combinations (50Hz/12kHz and 50Hz/16kHz). This field controls TN465B, TN2138, and TN2184 circuit packs. Available for outgoing or two-way trunk groups if PPM is enabled.
Valid Entry 12k 16k 50 50/12k 50/16k
Usage
The TN465B (or later) and TN2184 can only detect 12k and 16kHz PPM. Therefore, if 12k is administered, the circuit pack will be set to detect 12kHz.
The TN465B (or later) and TN2184 can only detect 12k and 16kHz PPM. Therefore, if 16k is administered, the circuit pack will be set to detect 16kHz.
The TN465B (or later) and TN2184 can only detect 12k and 16kHz PPM. Therefore, if 50 is administered, the circuit pack will be set to detect 16kHz.
The TN465B (or later) and TN2184 can only detect 12k and 16kHz PPM. Therefore, if 50/12k is administered, the circuit pack will be set to detect 12kHz.
The TN465B (or later) and TN2184 can only detect 12k and 16kHz PPM. Therefore, if 50/16k is administered, the circuit pack will be set to detect 16kHz.
Related links Direction on page 507 PPM on page 966
Make (msec)
The duration of the make interval (the pause between pulses) while the system is outpulsing digits using dial pulse signaling. The field cannot be blank.
Valid Entry
20 to 80 in increments of 5
10 to 40 in increments of 5
Usage
If the PPS (pulses per second) value is 10, the sum of the Make (msec) and Break (msec) values must equal 100.
If the PPS (pulses per second) value is 20, the sum of the Make (msec) and Break (msec) values must equal 50.
Related links Break (msec) on page 965 PPS on page 967
PPM
Determines if Periodical Pulse Metering (PPM) pulses should be collected from the public network to determine call cost. For CO, DIOD, FX, PCOL, and WATS trunks. Available only for outgoing or two-way trunk groups.
Related links Frequency on page 966 Direction on page 507
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Trunk Group: ATMS Thresholds
Direction on page 507
PPS
Available only for cama trunk groups.
Valid Entry 10 20
Usage
The rate (pulses per second) at which outgoing rotary pulses are sent over this trunk group.
The TN439, TN458, TN497, TN747Bv12 (or later), and TN767 circuit packs send rotary pulses at 10 pps only.
Related links Break (msec) on page 965 Make (msec) on page 966
Trunk Group: ATMS Thresholds
Available for outgoing or two-way trunk groups when ATMS is enabled for the system. Caution: Customers: Do not change fields on this page without assistance from Avaya or your network service provider.
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Trunk Group
Related links ATMS on page 830 Direction on page 507
Far-End Test No
The access number dialed to reach the terminating test line (TTL). Accepts up to 16 digits.
Trunk Contact
The name or telephone number of someone from the trunk vendor who can be contacted if there are problems with the trunks. Accepts up to 25 alphanumeric characters.
Trunk Length
This field is not required. Since noise on a trunk increases with the length of the trunk, however, this information might be useful.
Valid Entry 0 to 4 k 0 to 4 m
Usage The length of the trunk group in kilometers. The length of the trunk group in miles.
Trunk Vendor
The name of the vendor providing service over this trunk group that needs to be notified in the event of problems with the trunks in this trunk group. Accepts up to 22 alphanumeric characters.
TTL Contact
The name or telephone number of someone from the TTL vendor who can be contacted in the event of problems with the terminating test line. Accepts up to 25 alphanumeric characters.
TTL Type
Type of terminating test line (TTL) selected for testing trunks. The TTL type determines what ATMS tests can be completed and which threshold values need to be administered.
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Valid Entry 105-w-rl
105-wo-rl high-lts low-lts 100
102
Trunk Group: ATMS Thresholds
Usage 105 with return loss. Full range of 18 measurements or some defaults for return loss used (56A). 105 without return loss. Cannot return default values for far-end return loss. High-level tone source. Sends a fixed sequence of tones at 0 dBm. Low-level tone source. Sends a fixed sequence of tones at -16dBm. 100 type. Up to 5 measurements that sends a 1004 Hz tone, then a quiet termination. 102 type. One measurement that sends a 1004 Hz tone.
TTL Vendor
The name of the vendor supplying the terminating test line (TTL). Accepts up to 22 alphanumeric characters.
MARGINAL / UNACCEPTABLE
Allow ATMS Busyout, Error Logging and Alarming
Enables or disables ATMS error logging and alarming.
Marginal Threshold --Dev - 404 Hz Loss
Valid Entry 0 to 9
Usage
The maximum negative deviation of measured loss at 404 Hz from the 1004 Hz test tone noise level (in dB) allowed before reporting a trunk as out of tolerance. Smaller dB values are more restrictive.
Marginal Threshold -+Dev - 404 Hz Loss
Valid Entry 0 to 9
Usage
The maximum positive deviation of measured loss at 404 Hz from the 1004 Hz test tone loss level (in dB) allowed before reporting a trunk as out of tolerance. Smaller dB values are more restrictive.
Marginal Threshold --Dev - 2804 Hz
Valid Entry 0 to 9
Usage
The maximum negative deviation of measured loss at 2804 Hz from the 1004 Hz test tone loss level (in dB) allowed before reporting a trunk as out of tolerance. Smaller dB values are more restrictive.
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Trunk Group
Marginal Threshold -+Dev - 2804 Hz
Valid Entry 0 to 9
Usage
The maximum positive deviation of measured loss at 2804 Hz from the 1004 Hz test tone loss level (in dB) allowed before reporting a trunk as out of tolerance. Smaller dB values are more restrictive.
Marginal Threshold - Max - 1004 Hz Loss
Valid Entry 0 to 21
Usage
The maximum signal loss allowed for a 1004 Hz test tone (in dB) before a trunk is reported as out of tolerance. A smaller dB value is more restrictive.
Marginal Threshold - Maximum C Message Noise
Valid Entry 15 to 55
Usage
The maximum C-message noise telephone as measured within the voice band frequency range (500 to 2500 Hz) allowed before reporting a trunk as out of tolerance. Smaller values are more restrictive.
Marginal Threshold - Maximum C Notched Noise
Valid Entry 34 to 74
Usage
The maximum C-notched signal dependent noise interference in dBmC allowed before reporting a trunk as out of tolerance. Smaller values are more restrictive.
Marginal Threshold - Min -1004 Hz Loss
Valid Entry 2 to 21
Usage
The minimum signal loss allowed for a 1004 Hz test tone (in dB) before a trunk is reported as out of tolerance. A larger dB value is more restrictive.
Marginal Threshold - Minimum ERL
Valid Entry 0 to 40
Usage
The minimum low-frequency echo return loss in dB allowed before reporting a trunk as out of tolerance. Larger values are more restrictive.
Marginal Threshold - Minimum SRL-HI
Valid Entry 0 to 40
Usage
The minimum high-frequency signaling return loss in dB allowed before reporting a trunk as out of tolerance. Larger values are more restrictive.
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Trunk Group: ATMS Thresholds
Marginal Threshold - Minimum SRL-LO
Valid Entry 0 to 40
Usage
The minimum low-frequency signaling return loss in dB allowed before reporting a trunk as out of tolerance. Larger values are more restrictive.
Maximum Percentage of Trunks Which Can Be Removed From Service by ATMS
Available only if Allow ATMS Busyout, Error Logging, and Alarming are enabled.
Valid Entry 0, 25, 50, 75, 100
Usage
The highest percentage of trunks from the trunk group that can be removed from service at one time because of unacceptable transmission measurement results.
Related links Allow ATMS Busyout, Error Logging and Alarming on page 969
Unacceptable Threshold --Dev - 404 Hz
Valid Entry 0 to 9
Usage
The maximum negative deviation of measured loss at 404 Hz from the 1004 Hz test tone loss level (in dB) allowed before reporting a trunk as unacceptable. Smaller dB values are more restrictive.
Unacceptable Threshold -+Dev - 404 Hz
Valid Entry 0 to 9
Usage
The maximum positive deviation of measured loss at 404 Hz from the 1004 Hz test tone loss level (in dB) allowed before reporting a trunk as unacceptable. Smaller dB values are more restrictive.
Unacceptable Threshold --Dev - 2804 Hz
Valid Entry 0 to 9
Usage
The maximum negative deviation of measured loss at 2804 Hz from the 1004 Hz test tone loss level (in dB) allowed before reporting a trunk as unacceptable. Smaller dB values are more restrictive.
Unacceptable Threshold -+Dev - 2804 Hz
Valid Entry 0 to 9
Usage
The maximum positive deviation of measured loss at 2804 Hz from the 1004 Hz test tone loss level (in dB) allowed before reporting a trunk as unacceptable. Smaller dB values are more restrictive.
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Trunk Group
Unacceptable Threshold - Max - 1004 Hz Loss
Valid Entry 0 to 21
Usage
The maximum signal loss allowed for a 1004 Hz test tone (in dB) before a trunk is reported as unacceptable. A smaller dB value is more restrictive.
Unacceptable Threshold - Maximum C Message Noise
Valid Entry 15 to 55
Usage
The maximum C-message noise interference in dBmC above reference noise terminating on a telephone as measured within the voice band frequency range (500 to 2500 Hz) allowed before reporting a trunk as unacceptable. Smaller values are more restrictive.
Unacceptable Threshold - Maximum C Notched Noise
Valid Entry 34 to 74
Usage
The maximum C-notched signal dependent noise interference in dBmC allowed before reporting a trunk as unacceptable. Smaller values are more restrictive.
Unacceptable Threshold - Min - 1004 Hz Loss
Valid Entry 2 to 21
Usage
The minimum signal loss allowed for a 1004 Hz test tone (in dB) before a trunk is reported as unacceptable. A larger dB value is more restrictive.
Unacceptable Threshold - Minimum ERL
Valid Entry 0 to 40
Usage
The minimum low-frequency echo return loss in dB allowed before reporting a trunk as unacceptable. Larger values are more restrictive.
Unacceptable Threshold - Minimum SRL-HI
Valid Entry 0 to 40
Usage
The minimum high-frequency signaling return loss in dB allowed before reporting a trunk as unacceptable. Larger values are more restrictive.
Unacceptable Threshold - Minimum SRL-LO
Valid Entry 0 to 40
Usage
The minimum low-frequency signaling return loss in dB allowed before reporting a trunk as unacceptable. Larger values are more restrictive.
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Trunk Group: Group Member Assignments
Trunk Group: Group Member Assignments
For SIP trunks, the group member-assignment pages are not individually administered. The system automatically populates and displays these fields based on the number of members of SIP trunk groups previously specified.
Administered Members (min/max)
The minimum and maximum member numbers that have been administered for this trunk group.
Ans Delay
Caution:
Customers should not attempt to administer this field. Go to the Avaya Support website at http://support.avaya.com for current documentation, product notices, knowledge articles on administering this field.
Valid Entry 20 to 5100 in increments of 20
blank
Usage
The length of time the server running Communication Manager waits before it sends answer supervision for incoming calls on tie trunks using the TN722A or later, TN760 (B, C, or D), TN767, TN464 (any suffix), TN437, TN439, TN458, or TN2140 circuit packs.
Same as setting the field to zero.
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Trunk Group
Code
The type of circuit pack physically installed or logically administered at the location to which this member is assigned. If no circuit pack is installed or administered at the port address, the field is blank.
Mode
The signaling mode used on tie trunks with TN722A or later, TN760B or later, TN767, TN464 (any suffix), TN437, TN439, TN458, or TN2140 circuit packs. This entry must correspond to associated dip-switch settings on the circuit pack.
Caution:
Customers should not attempt to administer this field. Go to the Avaya Support website at http://support.avaya.com for current documentation, product notices, knowledge articles on administering this field.
Valid Entry e&m
simplex
Usage
For six-wire connections that pair two signaling wires with four voice wires. This configuration is used in the vast majority of systems in the United States.
For four-wire connections that do not use an additional signaling pair. This configuration is very rare in the United States.
Name
The name of the trunk group member. The name should identify the trunk unambiguously. Accepts up to 10 characters. Example
· The telephone number assigned to incoming trunks · The Trunk Circuit Identification number assigned by the service provider
Night
The night service destination for this trunk group member if different from the night service destination administered for the trunk group. Incoming calls are routed to this destination when the system is placed in night service mode.
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Trunk Group: Group Member Assignments
Valid Entry A valid extension attd blank
Usage The extension of the night destination for the trunk. Calls go to the attendant when night service is active. Not administered
Related links Night Service on page 911
Port
Valid Entry 1 to 64 A to E 0 to 20 01 to 04 (Analog TIE trunks) 01 to 31 1 to 250 V1 to V9 01 to 31
Usage First and second characters are the cabinet number. Third character is the carrier. Fourth and fifth characters are the slot number. Six and seventh characters are the circuit number.
Gateway Module Circuit
Sfx
The model suffix for the type of circuit pack physically installed at the location to which this member is assigned. If no circuit pack is installed at the port address, the field is blank.
Total Administered Members
The total number of members administered in the trunk group.
Type
Caution:
Customers should not attempt to administer this field. Go to the Avaya Support website at http://support.avaya.com for current documentation, product notices, knowledge articles on administering this field.
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Trunk Group
Valid Entry t1-stan t1-comp t5-rev type-5
Usage
Specifies the signaling type to be used with TN760B (or later release), TN722 (with any suffix), TN767, TN2140 (when the Trunk Type is cont), TN437, TN439, TN464 with any suffix, or TN458 circuit packs. Available only if the Trunk Type is blank or cont.
t5-rev is allowed only for the TN760D vintage 10 or later. When Type is t5 rev, Mode must be e&m.
Related links Trunk Type on page 936
Trunk Group: Protocol Variations
Available only for sip trunk groups.
Related links Group Type on page 508
Accept Redirect to Blank User Destination
The following table represents the way in which this field works.
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Valid entry y y
n n
Trunk Group: Protocol Variations
User destination in the Refer-To Result domain of the SIP request
present
Communication Manager places the call to the user destination.
missing
Communication Manager places the call over the signaling group specified in the Proxy Sel Rte Pat field of the Locations screen.
present
Communication Manager places the call to the user destination.
missing
Communication Manager rejects the SIP request.
Always Use re-INVITE for Display Updates
The system displays this field when the Group Type field is SIP.
Valid entry y
n
Usage
In SIP messages, Communication Manager sends the re-invite message to display update.
In SIP messages, Communication Manager sends the update message to display update. The default value is n.
Resend Display UPDATE Once on Receipt of 481 Response
The value in the Resend Display UPDATE Once on Receipt of 481 Response field determines whether Communication Manager should resend display update message on receiving 481 response to Session Manager, instead of dropping the call immediately.
When Communication Manager receives 481 as a response for display update message from Session Manager (only when the display update message is sent after the first non 100 response), then Communication Manager resends display update message to Session Manager.
Note:
The display update message must be after the first 18x response. If not, the display update message is not resent.
If Session Manager responds with 481 again, then Communication Manager drops the call. This field does not apply to all other update messages except display update message.
Valid Entry
Usage
Table continues...
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Trunk Group y n
If Communication Manager gets a 481 response for a display update message from Session Manager, Communication Manager resends the display update message once.
The Communication Manager does not re-send any update message (even if it is 481 response). By default, the value is n.
Block Sending Calling Party Location in INVITE
Based on the , Communication field value, Manager selects the IP address that must be included in the Via header in an outgoing call. The system displays this field if the value of the Group Type field in the trunk group screen is SIP.
Valid Entry y n
Usage
Communication Manager does not send the IP address of the caller in the INVITE.
Communication Manager sends an IP address that identifies the location of the caller. This IP address is included in the lower-most Via header in the INVITE. Based on the type of endpoint of the caller, Communication Manager determines the IP address to be sent.
· For calls made from H.323 endpoints, Communication Manager sends the IP address of the endpoint.
· For calls made from TDM endpoints or TDM trunks, Communication Manager sends the IP address of the media processor used in the call. This value is the default value.
Build Refer-To URI of REFER From Contact For NCR
The system displays this field only for SIP trunk groups for which Network Call Redirection is enabled. Communication Manager uses the REFER message during a call transfer.
Valid entry y
n
Usage
Communication Manager uses the address in the Contact header to create the URI of the Refer-To header in a REFER message.
Communication Manager uses the address in the From header to create the URI of the Refer-To header in a REFER message.
Convert 180 to 183 for Early Media
It is used for early media and direct media cut-through. When SDP answer is returned by Communication Manager in 18x messages, some entities have a problem receiving the SDP in a 180 Ringing message. If the Convert 180 to 183 for Early Media field is set to y, Communication Manager puts the SDP into a 183 Session Progress message. The default value is n.
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Trunk Group: Protocol Variations
Enable Q-SIP
The system displays Enable Q-SIP, only when the Group Type field is sip.
Valid Entry y
n
Usage
Enables the QSIG over SIP (Q-SIP) feature for the trunk group. If trunk members are already assigned to this trunk group, you must not change the value of Enable QSIP field. If you change the value of Enable Q-SIP field, the system displays an error message and prompts you to remove all assigned members before enabling Q-SIP.
The QSIG over SIP feature for the trunk group is disabled. By default, the value is n.
Identity for Calling Party Display
The system displays this field, when the Group Type field is sip.
This field determines which header to retrieve display information for the calling party when both the From and P-Asserted-Identity headers are available.
Valid Entry P-Asserted-Identity
From
Usage
If a call is terminating at SIP, H.323, or DCP stations, or at outgoing H.323 or SIP trunk, Communication Manager displays the name and Calling Party number from the PAI header instead of the FROM header. This is the default value.
If a call is terminating at SIP, H.323, or DCP stations, or at outgoing H.323 or SIP trunk, Communication Manager displays the name and Calling Party number from the FROM header instead of the PAI header.
Interworking of ISDN Clearing with In-Band Tones
The setting of this field is for call-clearing of the tandem calls in which the incoming trunk to Communication Manager is a SIP trunk and the outgoing trunk from Communication Manager is an ISDN trunk. The far end at the ISDN PRI trunk ends the call.
If you select the keep-channel-active option, Communication Manager keeps the call active so that the caller can hear an in-band tone or announcement played over the ISDN trunk.
If you select the drop-with-sip-error option, Communication Manager detects the cause of call clearing, maps ISDN Cause Value to a SIP error response message, and sends the response message to the caller. With the drop-with-sip-error option, an adjunct placing outgoing calls is informed that the call has ended and the user can disconnect the call immediately instead of waiting for the ISDN network to end the call after the tone or announcement is complete.
Note:
An in-band tone is an announcement or a tone that provides call-clearing information.
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Trunk Group
Valid entry keep-channel-active
drop-with-sip-error
Usage
Communication Manager sends a SIP 183 Progress message to the caller. The ISDN network plays an in-band tone that the caller hears.
The default value is keep-channel-active.
Communication Manager sends a SIP error response message, which specifies the cause of call-clearing to the caller. The called party plays an in-band tone. The caller does not hear the in-band tone or announcement.
Mark Users as Phone
Enables or disables the encoding of URIs in call control signaling messages originated at the gateway with the "user=phone" parameter. No subscription messages are encoded with the "user=phone" parameter, even when the field is set to y. Default is n.
Note: Do not change the default of n for this field unless you are sure that every recipient of SIP calls using this trunk can accept and properly interpret the optional "user=phone" parameter. Enterprise users without support for "user=phone" in their SIP endpoints will experience adverse effects, including rejected calls.
Network Call Redirection
If enabled, Network Call Redirection (NCR) service is signaled over this trunk group. NCR only works on trunk groups connected to Service Providers that support NCR.
Prepend '+' to Calling/Alerting/Diverting/Connected Number
The system displays this field only for SIP trunk groups. If you set this field to y, when a call is routed over a trunk group, Communication Manager inserts a leading plus (+) sign, if absent, in outgoing calling, alerting, diverting, or connected numbers. The default value is n. To prevent Communication Manager from inserting a leading plus (+) sign in the calling number when a call is routed from Communication Manager to Session Manager, set the Numbering Format field to private and the Prepend '+' to Calling/Alerting/Diverting/Connected Number field to n. Related links Group Type on page 508
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Trunk Group: Protocol Variations
QSIG Reference Trunk Group
The system displays this, only when the Group Type field is sip and the Enable Q-SIP is set to y.
Valid Entry 1 to 2000
1 to 99 blank
Usage
Assigns a number for the QSIG trunk group. If trunk members are already assigned to this trunk group, you must not change the value of QSIG Reference Trunk Group field. If you change the value of this field, the system displays an error message and prompts you to remove all assigned members before enabling Q-SIP. Applicable for Avaya supported servers.
For the Avaya supported servers, see Avaya Aura® Communication Manager Hardware Description and Reference.
(For Avaya S8300E Server) Assigns a number for the QSIG trunk group. If trunk members are already assigned to this trunk group, you must not change the value of QSIG Reference Trunk Group field. If you change the value of this field, the system displays an error message and prompts you to remove all assigned members before enabling Q-SIP.
No QSIG trunk group is assigned. By default, the value is blank.
Request URI Contents
Use this field to configure Communication Manager to allow or restrict the called-party digits in the Request URI of the SIP INVITE message, REFER message, or 3xx redirect response for INVITE message.
Valid entry may-have-extra-digits
called number-only
Usage
Communication Manager routes an incoming Request URI without considering the total number of digits. Communication Manager might route the call using an entry that matches fewer digits than the total number in the Request URI.
Communication Manager considers the total number of digits when routing an incoming Request URI. The routing fails if a match that incorporates all digits is unavailable.
Send Diversion Header
This field is available only when the Service Type field is set to public-ntwrk. If you enable the Send Diversion Header field, the SIP diversion header is sent on the public SIP trunks. This header allows service providers to use the ten digit number of the forwarding party for functions
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Trunk Group
such as billing. Customers can administratively enable or disable use of the SIP diversion header on any public SIP trunk. The default value is n. Related links Service Type on page 512
Send Transferring Party Information
Enables or disables sending the transferring party information on a transferred call. Default is disabled.
Support Request History
You can activate the Support Request History field for a SIP trunk to capture the request location that was sent to a proxy. This information is used to send diversion information and notify the origin of the call to the far end equipment. If this field is disabled, History-Info header is not included in any requests.
Note:
This feature does not apply to Off PBX SIP station trunk groups for which the Communication Manager never suppresses the History-Info header.
Telephone Event Payload Type
Valid Entry 96 to 127 blank
Usage
The default payload type offered by Communication Manager for SIP trunks. The payload type number encoding for originating (offering) the RFC 2833 RTP "telephone-event" payload format is based on the administered number from this field. This value is used only for Communication Manager originations (outgoing offers).
Default is 127.
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Chapter 148: Uniform Dial Plan Table
The UDP provides a common dial plan length -- or a combination of extension lengths -- that can be shared among a group of Avaya servers. Additionally, UDP can be used alone to provide uniform dialing between two or more private switching systems without ETN, DCS, or Main/Satellite/ Tributary configurations. Available only if Uniform Dialing Plan is enabled for the system. Example command: change uniform-dialplan n, where n is the uniform dial plan length.
Related links Uniform Dialing Plan on page 841
Conv
Enables or disables additional digit conversion.
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Uniform Dial Plan Table
Del
Valid Entry 0 to 9
Usage
The number of digits to delete before routing the call. This number must be less than or equal to the number of user dialed digits the system collects to match to the matching pattern.
Related links Len on page 984
Insert Digits
Use this field to specify the digits or the number of digits of the location prefix that Communication Manager inserts in the dialed string before routing a call.
Valid entry 0 to 9
Lx
Usage
The digits that replace the deleted portion of the dialed number. The deleted portion is the number of digits that Communication Manager deletes specified in the Del field. You can enter up to 10 digits.
Leave the field blank to delete the digits specified in the Del field.
The variable x is a number from 1 to 11 and represents the number of leading digits of the administered location prefix. Communication Manager prepends the digits of the location prefix to the dialed string. The value of x must be less than the value of the Prefix field on the Location screen.
Related links Locations on page 564
Len
Valid Entry 1 to 18
Usage
The number of user-dialed digits the system collects to match to this Matching Pattern. This number must be greater than or equal to the Matching Pattern value. The value 2 can be used only when Insert Digits contains an Lx value.
Related links Insert Digits on page 984 Matching Pattern on page 985
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Matching Pattern
Matching Pattern
Communication Manager matches the value of the Matching Pattern field to the dialed numbers.
Valid entries 0 to 9
blank
Usage Communication Manager matches the value with the dialed numbers. You can enter up to 16 digits. Leave the field blank if you do not want to specify a pattern.
Net
Valid Entry aar ars enp ext
Usage
The server or switch network used to analyze the converted number. The converted digit-string is routed either as an extension number or through its converted AAR address, its converted ARS address, or its ENP node number.
For enp, the ENP node number must be administered, digit conversion must be disabled, and the Insert Digits must not be administered.
Related links Conv on page 983 Insert Digits on page 984 Node Num on page 985
Node Num
Valid Entry 1 to 999
Usage The ENP (Extension Number Portability) Node Number.
Percent Full
Displays the percentage (0 to 100) of the memory resources allocated for the uniform dial plan data that are currently being used. For details on the system memory resources, see Avaya Aura® Communication Manager System Capacities Table.
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Chapter 149: User Profile
This screen is described in Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and Servers. For more information on administering user profiles and logins, see AAA Services in Avaya Aura® Communication Manager Feature Description and Implementation.
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Chapter 150: Variables for Vectors
This screen creates variables and defines the necessary parameters for each variable type. Specifies the variable type, the name to use for the variable, the size of the variable, how the variable gets set or assigned, and whether the variable is local or global. Up to 702 variables can be supported using A to Z and AA to ZZ rows. Example command: change variables
Assignment
A number to pre-assign to the variable. This field displays the current value for global values. Administration is optional. Available only when the type is value or collect G.
Related links Type on page 988
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Variables for Vectors
Description
An identifying name or description of the vector variable. Administration is optional. Default is blank. Accepts up to 27 characters.
Length
Valid Entry 1 to 16
Usage
The maximum number of digits from the data to assign to the variable. This field does not apply to the doy, dow, and tod variables. This entry is required for all types to which it applies.
Scope
Valid Entry G L
Usage The variable is used globally. The variable is used locally.
Start
Valid Entry 1 to 96
Usage
Specifies the beginning character position of the data digits string to be used for assigning to the variable. The combination of the Start position and maximum length of the digits string defines what is to be assigned to the variable. If the number of digits to be used is less than the maximum length specified, only that portion is assigned to the variable.
Type
Use this field to specify the vector variable type. The following are the valid entries: · ani · asaiuui · collect
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VAC
· dow · doy · stepcnt · tod · value · vdn · vdntime · agent
VAC
Valid Entry 1 to 9 blank
Usage The Vector Variable Feature Access Code (FAC) to use for changing the value.
Var
Valid Entry A to Z, AA to ZZ
Usage The letter identifying the row of a specific vector variable.
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Chapter 151: Vector Directory Number
Defines vector directory numbers (VDN) for the Call Vectoring feature. A VDN is an extension number used to access a call vector. Each VDN is mapped to one call vector. VDNs are software extension numbers (that is, not assigned to physical equipment). A VDN is accessed through direct dial local telephone company central office trunks mapped to the VDN (incoming destination or night service extension), DID trunks, and LDN calls. The VDN can be the Night Destination for a LDN. An asterisk (*) next to a field indicates that the field follows VDN override rules when the system changes the active VDN for a call. Example command: add vdn n, where n is the VDN extension number.
Vector Directory Numbers: page 1
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Vector Directory Numbers: page 1
1st/2nd/3rd Skill
This field follows VDN override rules when the system changes the active VDN for a call. Available only if Expert Agent Selection (EAS) is enabled and Meet-me Conferencing is disabled.
Valid entry 1 to 8000 blank
Usage The skill number for each skill. Not administered. This is the default.
Related links Expert Agent Selection (EAS) on page 843 Meet-me Conference on page 993
Acceptable Service Level (sec)
Available only if BCMS/VuStats Service Level is enabled for the system and the VDN is measured by BCMS.
Valid Entry 0 to 9999
blank
Usage
The number of seconds within which calls to this VDN should be answered. This allows BCMS to report the percentage of calls that were answered within the specified time.
Not administered. This is the default.
Related links BCMS/VuStats Service Level on page 842 Measured on page 992
Allow VDN Override
Available only if Meet-me Conferencing is disabled for that particular VDN
Valid Entry y
n
Usage
Allows the system to change the routed-to VDN to the "active" VDN for a call. The "active" VDN is the VDN to be used for parameters associated with the call such as VDN name, skills, tenant number, BSR application, VDN variables, and so on. The field follows VDN override rules when the system changes the "active" VDN for a call.
The routed-to VDN does not become the active VDN. The parameters of the original VDN are used. This is the default.
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Vector Directory Number
Related links Meet-me Conference on page 993
Attendant Vectoring
Indicates whether or not Attendant Vectoring is selected on this VDN. Attendant Vectoring does not support Call Center features.
Related links Attendant Vectoring on page 830
COR
Valid Entry 0 to 995
Usage
The class of restriction (COR) of the VDN consisting of a one- or two-digit number. This field cannot be blank.
Destination
Specifies if the calls are routed using a Vector Number or Policy Routing Table.
Extension
The extension number of the VDN. The extension is a number that starts with a valid first digit and length as defined by the system's dial plan.
Measured
Enables measurement data collection for this VDN. Data can be collected for reporting by BCMS or the Call Management System. Available only if Meet-me Conferencing is disabled. BCMS must be enabled for the system for this field to be set to internal or both. In addition, the appropriate CMS release must be administered for the system if this field is external or both.
Valid Entry internal external both none
Usage Data is measured internally by BCMS. Data is measured internally by CMS. Data is measured internally by both BCMS and CMS. Data is not measured. This is the default.
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Related links Meet-me Conference on page 993 BCMS (Basic) on page 842
Vector Directory Numbers: page 1
Meet-me Conference
Enables or disables Meet-me Conferencing for this VDN. Determines if the VDN is a Meet-me Conference VDN.
Note:
If the VDN extension is part of your DID block, external users are able to access the conference VDN. If the VDN extension is not part of the DID block, only internal callers on the network (including DCS or QSIG) or remote access callers can access the conference VDN.
Available only if Enhanced Conferencing is enabled for the system. If Enhanced Conferencing is enabled, but no other vectoring options are enabled, only Meet-me Conference vectors can be assigned.
Related links Enhanced Conferencing on page 833
Name
The name associated with the VDN. When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call. Accepts up to 27 alphanumeric characters.
The name might be truncated on agents' displays depending on the application. When information is forwarded with an interflowed call, only the first 15 characters are sent.
Note:
Supported by Unicode language display for the 4610SW, 4620SW, 4621SW, and 4622SW, 1600-Series, 9600-Series IP Deskphones, 9600-Series H.323 Deskphones, and Avaya J100 Series IP Phones. Unicode is also an option for the 2420J telephone when the Display Character Set is katakana. For more information about 2420J, see 2420 Digital Telephone User's Guide.
Related links Display Character Set on page 821 Meet-me Conference on page 993
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Vector Directory Number
Number
Valid Entry 1 to 8000
Usage
The number of the vector or the Policy Routing Table (PRT) to which the calls are allocated.
Service Objective
Use this field to assign a service level to the VDN. The system displays this field only if you set the Dynamic Advocate field to y. The range of the acceptable value is between 1 to 9999. The default value is 20. For more information on Service Objective, see Avaya Business Advocate User Guide.
Related links BCMS/VuStats Service Level on page 842 Dynamic Advocate on page 843
SIP URI
Use this field to place a call using the URI. You can enter up to 70 characters. You can use letters, numbers, and special characters. Following are not supported:
· Unicode characters · Eurofont characters · Using same URIs across hunt groups, VDNs, and stations within the same Communication
Manager. Example support-helpdesk@example.com
TN
Use this field to specify a tenant number. You can enter a value from 1 to 250.
Unicode Name
This read-only field shows the assignment Unicode Name to the VDN.
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Vector Directory Numbers: page 1
Valid entry n y
Usage Unicode Name is not assigned to the VDN. Unicode Name is assigned to the VDN.
For more information on Unicode Language Display, see Administering Avaya Aura® Communication Manager.
VDN of Origin Annc. Extension
The extension number of the VDN of Origin announcement. A VDN of Origin announcement is a short recording that identifies something about the call originating from the VDN. The agent hears the recording just prior to the delivery of the call. This field follows VDN override rules when the system changes the "active" VDN for a call.
Available only if VDN of Origin Announcement is enabled for the system and Meet-me Conferencing is disabled.
Related links Meet-me Conference on page 993 VDN of Origin Announcement on page 847
Vector Number
An identifying number that specifies the call vector that is accessed through the VDN. This field cannot be blank.
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Vector Directory Number
Vector Directory Number: page 2 (Meet-me Conference disabled)
Related links Meet-me Conference on page 993
After Xfer or Held Call Drops
Use this field to put an auto-in agent into the Timed ACW mode when an ACD or DAC call on hold drops or the call is transferred and the Timed ACW feature is active. This replaces the Available mode.
Valid Entry y
n
Usage
Communication Manager places an agent in the Timed ACW state if a caller on hold disconnects or the call is transferred, in addition to the existing conditions for Timed ACW.
Communication Manager continues with the existing operation. This is the default value.
AUDIX Name
The name of the AUDIX machine if this VDN is associated with the AUDIX vector.
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Vector Directory Number: page 2 (Meet-me Conference disabled)
BSR Application
Available only if Lookahead Interflow (LAI) and Vectoring (Best Service Routing) are enabled for the system. When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call.
Valid Entry 1 to 511 blank
Usage
A one- to three-digit number that specifies an application plan for the VDN when using multi-site Best Service Routing with this VDN.
Related links Lookahead Interflow (LAI) on page 844 Vectoring (Best Service Routing) on page 845
BSR Available Agent Strategy
This field determines how Best Service Routing (BSR) identifies the best split or skill in an agent surplus situation to route a call. The system displays this field only if Vectoring (Best Service Routing) is enabled. If Meet-me Conferencing is disabled, the BSR Available Agent Strategy field follows VDN override rules when the system changes the active VDN for a call.
Valid entry 1st-found UCD-LOA
UCD-MIA
EAD-LOA
EAD-MIA
Usage
The call is routed to the first available agent.
The call is routed to the least occupied agent, without regard to skill level. This option is available only if Least Occupied Agent (LOA) or Business Advocate is enabled.
The call is routed to the most idle agent, without regard to skill level. This option ensures a high degree of equity in agent workloads even when call-handling times vary.
The call is routed to the highest skill level agent with the lowest occupancy. This option is available only if Expert Agent Selection, and Least Occupied Agent (LOA) or Business Advocate are enabled.
The call is routed to the most idle agent with the highest skill level. This option is available only if Expert Agent Selection (EAS) is enabled.
Related links Business Advocate on page 842 Expert Agent Selection (EAS) on page 843 Least Occupied Agent on page 844 Vectoring (Best Service Routing) on page 845
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Vector Directory Number
BSR Local Treatment
Enables or disables local server control of BSR treatment. In a multi-site BSR configuration, a call that arrives at a local communication server can be rerouted to a remote server located in a different part of the world. This feature maintains control at the local server and allows the server to provide local audio feedback for IP and ISDN calls, or to take back the call while the call waits in queue on a remote server. When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call.
Note:
This field must be enabled on both the local and remote vdns, or else call interflow attempts might result in dropped calls.
BSR Tie Strategy
Available only if Vectoring (Best Service Routing) is enabled for the system. When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call.
Valid Entry system 1st-found
alternate
Usage
System-wide settings apply.
BSR uses the previously selected best choice as the best skill or location. This is the default setting.
Alternates the BSR selection algorithm when a tie in EWT or available agent criteria occurs. Every other time a tie occurs for calls from the same VDN, the step with the tie is selected to send the call instead of the first selected split, skill, or location. This helps balance the routing when the cost of routing remotely is not a concern.
Related links Vectoring (Best Service Routing) on page 845
Call Origin
The Call Origin field is applicable only if you enter a VDN extension in the Return Destination field. Use this field to administer the types of calls that will be redirected to the extension in the Return Destination field when the destination is released.
Valid Entry external
Usage
Applies VDN Return Destination only to incoming external trunk calls directly to the VDN. This is the default setting.
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Valid Entry internal
both
Vector Directory Number: page 2 (Meet-me Conference disabled)
Usage Applies VDN Return Destination only to internal calls that have not been through vector processing. Internal calls for this purpose include trunk calls that are transferred, forwarded, or covered to the VDN or ASAI routed. Applies VDN Return Destination to both internal and external calls.
Display VDN for Route-To DAC
Enables or disables the display of the VDN when a call arrives as a result of a route-to at an agent terminal under the following conditions:
· A route-to number with coverage is enabled, or route-to digits with coverage is enabled and the vector command routes a call to an agent as an EAS direct agent call.
· Adjunct routing routes a direct agent call to the agent.
When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call.
Available only if Expert Agent Selection (EAS) is enabled for the system.
Related links Expert Agent Selection (EAS) on page 843
Observe on Agent Answer
Enables or disables a service observer to start observing a call to the VDN when the call is delivered to the agent or station. The ability to observe VDNs when the call is delivered to an agent/station is activated by setting the Observe on Agent Answer field to y. The observer sees the name of the VDN, agent, or trunk as each is accessed in sequence by the VDN. For example, during vector processing the VDN name is displayed, but when the call connects to an agent, the agent name is displayed. When the observer connects to a call in vector processing, the system maintains the connection until the call is disconnected or the observer disconnects, even if the call is routed or transferred externally. If the observer does not disconnect after one observed call is disconnected, the observer is connected to another call on the same VDN. Observing is listen only as long as the call is in vector processing. Once the call is out of vector processing, an observer with listen/talk capability can talk as well as listen.
Note:
When the Observe on Agent Answer field is set to y, service observers of the VDN are not connected to a call until the call is delivered to an agent. The default setting of this field is n. With the default setting, the non-by location observer of the VDN can hear the call throughout the vector processing and delivery to the agent. For information on Observe on Agent Answer, see Avaya Aura® Call Center Elite Feature Reference.
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Vector Directory Number
Pass Prefixed CPN to VDN/Vector
This feature is used to optionally pass the CPN prefix to VDNs and vectors as part of the CPN. This option can be set on a system wide basis, and overridden, if required, on an individual VDN screen. Accordingly, international call handling can be improved by prioritizing and routing more intelligently, such as by automatically routing calls to agents who speak the pertinent language. For example, if a customer from Germany calls in to a call center in Spain, based on the international code of Germany (0049 - international prefix + country code), the call can be routed to a German speaking agent. Also, an international call might be given priority over a national call to save toll charges.
PC Predictive Reports Skill
The PC Predictive Reports Skill field is applicable only if the ASAI Link Plus Capabilities or the Computer Telephony Adjunct Links field is enabled on page 3 of the System-Parameters Customer-Options screen, and if you enable the Reporting for PC or POM Calls field on page 2 of the Vector Directory Number screen.
Valid Entry 1 to 99 1 to 8000 blank
Usage Skill number for S8300E switches. VDN option applies to PC reporting. Skill number for all other platforms. VDN option applies to PC reporting. VDN option applies to Proactive Outreach Manager (POM) calls.
Related links Reporting for PC or POM Calls on page 1000 ASAI Link Plus Capabilities on page 829 Computer Telephony Adjunct Links on page 831
Reporting for PC or POM Calls
The Reporting for PC or POM Calls field indicates that the VDN is used for POM or Proactive Contact (PC) outbound calling. If you have switch-classified outbound calling with PC, use this field and the associated skill field to activate the improved integration with PC feature. If you have POM, use the field to activate this VDN for indication to reporting of a Proactive Outreach Manager call to the VDN. Valid entries are y and n. The default is n.
Note:
You can administer the field only after you set the Measured field on page 1 of the Vector Directory Number screen to either external or both.
For more information on how to use the feature, see Proactive Contact outbound calling improved reporting in the Avaya Aura® Call Center Feature Reference document.
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Related links Measured on page 992
Vector Directory Number: page 2 (Meet-me Conference disabled)
Return Destination
The VDN extension number to which an incoming trunk call is routed if it returns to vector processing after the agent drops the call. When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call.
Send VDN as Called Ringing Name Over QSIG
Valid Entry y
n
Usage
If the ISDN trunk call is made, receiver can view the VDN name when the telephone is ringing.
The receiver cannot view the VDN name when the telephone is ringing. The default value is n.
Used for BSR Polling
Use the Used for BSR Polling field to prevent Communication Manager from sending messages to CMS or Avaya IQ regarding unmeasured calls for unmeasured VDNs that route to vectors with the BSR reply-best command.
The Used for BSR Polling field is applicable to unmeasured VDNs assigned to a vector that has consider steps terminated by the reply-best step and no-queuing step. The trunk groups that are used to route to the VDNs are also unmeasured.
For Used for BSR Polling to function, set the Measured field to none.
Valid entry y
n
Usage
Communication Manager does not send messages for unmeasured calls to CMS or Avaya IQ.
Communication Manager sends messages, even for unmeasured calls, to CMS or Avaya IQ. The default value is n.
VDN Override for ASAI Messages
Determines if the active VDN is sent as the called number for ISDN Trunk ASAI messages. When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call. The "active" VDN is the VDN receiving the call and will be
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Vector Directory Number
changed to a routed-to VDN if Allow VDN Override is enabled. Available only if ASAI Link Core Capabilities is enabled for the system.
Valid Entry n
ISDN Trunk all
Usage
The "Called Number" information is sent for the "Call Offered", "Alerting", "Queued", and "Connect" ASAI event notification messages. The adjunct-request message is always the called VDN extension in the Called Number IE sent in the incoming ISDN SETUP message or the local call's called number and does not change after routing to the called VDN and subsequent routed-to VDNs.
When an incoming ISDN trunk call is routed to this VDN, the "Called Number" information sent in the ASAI event and "Adjunct Route Request" ASAI messages is the "active VDN" extension. This extension becomes associated with the call based on the VDN Override rules. This option does not apply to local/internal calls.
The active VDN is used for the called number for all types of calls to the VDN, including local/internal calls as well as external incoming ISDN trunk calls.
Related links Allow VDN Override on page 991 ASAI Link Core Capabilities on page 829
VDN Timed ACW Interval
Valid Entry 1 to 9999 blank
Usage
Sets the length of the timed ACW Interval. When a value is entered in this field, an agent in auto-in work mode who receives a call from this VDN is automatically placed into After Call Work (ACW) when the call drops. When the administered time is over, the agent automatically becomes available. This field takes precedence over the hunt group Timed ACW Interval. When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call.
Related links Timed ACW Interval (sec) on page 407
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Vector Directory Number: page 2 (Meet-me Conference enabled)
Vector Directory Number: page 2 (Meet-me Conference enabled)
Related links Meet-me Conference on page 993
Conference Access Code
Security alert: To ensure conference security, always assign an access code to a Meet-me Conference VDN. The Meet-Me Conference VDN access code. Accepts up to six digits.
Conference Controller
Controls which user is allowed to change the access code for a Meet-me Conference VDN using a feature access code. This can be a local user or someone dialing in from remote access trunks.
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Vector Directory Number
Valid Entry Extension number
blank
Usage
Only a user at this extension can change the access code for that VDN using a feature access code.
Any station user that is assigned with console permissions can change the access code for that VDN using a feature access code.
Conference Type
The conference type that is appropriate for the call.
Valid Entry local
expanded
Usage For six or fewer participants. Default is local.
Prior to Release 8.1.2, valid entry was 6-party. In Release 8.1.2, the valid entry is changed to local.
Note:
For 12 parties to participate in a conference, you must enable the 12party Conferences field in the Feature-Related System-Parameters screen. Enables the Expanded Meet-me Conference feature. For a conference with more than six participants.
Route-to Number
The ARS or AAR Feature Access Code (FAC) followed by the routing digits. Alternately, the unique UDP extension. Accepts up to 16 digits. Allows administration of the routing digits (the ARS/AAR Feature Access Code with the routing digits and the Conference ID digits for the VDN). Available only for expanded conference types.
Note:
The Route-to Number must be unique across all Expanded Meet-me Conference VDNs.
Related links Conference Type on page 1004
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Vector Directory Number: page 3
Vector Directory Number: page 3
VDN VARIABLES
When Meet-me Conferencing is disabled, the system displays an asterisk (*) next to the heading, indicating that variables V1 through V9 follow VDN override rules when the system changes the "active" VDN for a call.
Related links Meet-me Conference on page 993
Assignment
Valid Entry 0 to 9
blank
Usage
Assigns an up to 16-digit unvalidated decimal number to each of the VDN variables V1 through V5.
No decimal number is assigned to the VDN variable.
Description
A description of the VDN variable. Accepts up to 15 characters.
Var
The number assigned to the VDN variable.
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Vector Directory Number
Apply Ringback for Auto Answer calls
The Apply Ringback for Auto Answer calls field is applicable only if you enable the Auto Answer field on either page 2 of the Station screen or on page 1 of the Agent Login ID screen. You can use the field to not apply ringback to the callers for calls delivered to agents in the auto answer mode when application of ringback is undesirable. This field follows VDN Override rules and the setting on the active VDN for the call is applicable.
Valid Entry y
n
Usage
Applies audible ringback to the callers during any delay experienced for calls delivered to agents in the auto answer mode. This is the default value.
Prevents returning audible ringback to the callers for calls delivered to agents in the auto answer mode.
Related links Auto Answer on page 46 Auto Answer on page 765
Daylight Saving Rule
Defines the daylight saving time rule. The daylight saving time rule is applied to goto time-ofday commands in the vector that is assigned to the VDN. The time-of-day calculations are based on the local time of the receiving call's VDN. The assigned rule number applies start and stop rules that are administered for the system for that rule.
Valid Entry system 0
1 to 15
Usage
The system uses the same daylight saving time rule as the system clock.
No daylight saving rule is applied. If the system time has a daylight saving rule specified, this rule is removed before evaluating the goto if time-of-day conditional.
The Daylight Saving Rule number. When you use a number other than 0, the rule associated with the main server clock display time and the main server offset are not used. The offset and rule assigned to the active VDN for the call are applied to the operating system standard time so that local time for the VDN is used to test the time-of-day step.
Related links Daylight Saving Rules on page 204 VDN Time Zone Offset on page 1007
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Vector Directory Number: page 3
Use VDN Time Zone For Holiday Vectoring
Valid Entry y
n
Usage
Communication Manager uses the VDN time zone offset and DST adjustment for processing the date and time assigned in the Holiday Table screen.
Communication Manager uses the system time for processing the date and time assigned in the Holiday Table screen. This is the default value.
VDN Time Zone Offset
Valid Entry +, 00-23 - hour 00-59 - minute
Usage
This field is applied against the switch clock when a time of day vector command is executed. Daylight saving time changes are handled by the switch clock using the existing operation.
When Meet-me Conferencing is disabled, this field follows VDN override rules when the system changes the "active" VDN for a call.
The valid entries are based on a syntax of +HH:MM. The default is +00:00. When the default is set, the system switch time is used without modification.
Related links Meet-me Conference on page 993
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Chapter 152: Video Bridge
Used to configure available ad-hoc conferencing resources. For more detailed information on Avaya Video Telephony, see Avaya Video Telephony Solution Networking Guide, 16-601423. Example command: add video-bridge n, where n is the video bridge number.
Bridge ID
The ID number for this video bridge.
Call Rate
The maximum allowable call rate for the conference. Available only when Far End Resource Info is disabled and Type is not administered.
Related links Far End Resource Info on page 1008 Type on page 1010
Far End Resource Info
Enables or disables far end resource tracking and reporting.
Valid Entry y n
Usage The far end tracks port usage and provides updates on resource availability. No resource information is provided from the far end.
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ID Range Start/End
ID Range Start/End
A range of conference IDs that this video bridge can use. There must be enough IDs so that all ports can be used -- one ID for every six ports. Accepts up to nine digits. The default is blank. Available only for h.323 trunk groups. Related links Group Type on page 508
Max Ports
Valid Entry 3 to system max
Usage The maximum number of video conferencing ports for this video bridge. Default is none.
Name
The name that identifies this video bridge. Accepts up to 30 alphanumeric characters.
Priority Factory Number
Priority versus Standard factory number depends on who creates the conference; if a user with Priority Video permissions creates the conference, the Priority Factory Number is used. Priority Factory Number can have a dedicated video bridge or a bridge with bandwidth more than the Standard Factory Number. Available only for h.323 or sip trunk groups when the Far End Resource Info is enabled.
Valid Entry 1 to 9 digits (0,9)
blank
Usage
At least one of Priority Factory Number or Standard Factory Number must be filled in. Standard and Priority factory numbers can be the same.
Priority calls can use the bridge, but prefer a bridge with a priority factory. This is the default.
Related links Far End Resource Info on page 1008 Group Type on page 508
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Video Bridge Standard Factory Number on page 1010
Standard Factory Number
Priority versus Standard factory number depends on who creates the conference; if a user with Priority Video permissions creates the conference, the Priority Factory Number is used. Priority Factory Number has a dedicated video bridge or a bridge with better bandwidth than theStandard Factory Number. Available only for h.323 or sip trunk groups. For h.323, the Far End Resource Info must be enabled.
Valid Entry 1 to 9 digits (0,9)
blank
Usage
At least one of Priority Factory Number or Standard Factory Number must be filled in. Standard and Priority factory numbers can be the same.
Non-priority conferences are unable to use this video bridge. Default is blank.
Related links Priority Factory Number on page 1009 Far End Resource Info on page 1008 Group Type on page 508
Trunk Groups
Valid Entry 1 to 2000 blank
Usage
Assigns trunk groups to this video bridge. At least one incoming and one outgoing trunk, or a two-way trunk, must be assigned. All trunks on a given video bridge must be the same type; H.323 and SIP trunks cannot be mixed.
The default is blank.
Related links Group Type on page 508
Type
The type of video bridge. Can be administered for existing or new video bridges. Available only if Far End Resource Info is disabled.
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Type
Valid Entry Exact
Maximum
Region
Usage
All participants in the conference must use the exact rate specified or be in the audio-only mode.
Call Rate is enabled, and any call rate up to the configured rate is allowed in a conference. Default is Maximum.
The initial entry for an existing video bridge. Call Rate is unavailable.
Related links Call Rate on page 1008 Far End Resource Info on page 1008
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Chapter 153: Virtual MAC Addresses
Lists the virtual Media Access Control (MAC) addresses on the system. Example command: display virtual-mac-address n, where n is the virtual MAC addresses table number.
MAC Address
Virtual MAC address shared by duplicated TN2602AP circuit packs. Accepts up to 15 alphanumeric characters.
Note: The 4606, 4612, and 4624 telephones do not support the bearer duplication feature of the TN2602AP circuit pack. If these telephones are used while an interchange from archive to standby media processor is in process, calls might be dropped.
Used
Indicates whether or not the associated virtual MAC address has been assigned in the system.
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Chapter 154: Resources
Communication Manager documentation
The following table lists the documents related to Communication Manager. Download the documents from the Avaya Support website at http://support.avaya.com.
Title
Description
Design
Avaya Aura® Communication
Provides an overview of the features of
Manager Overview and Specification Communication Manager
Avaya Aura® Communication Manager Security Design
Describes security-related issues and security features of Communication Manager.
Avaya Aura® Communication Manager System Capacities Table
Describes the system capacities for Avaya Aura® Communication Manager.
LED Descriptions for Avaya Aura® Communication Manager Hardware Components
Describes the LED for hardware components of Avaya Aura®
Communication Manager.
Avaya Aura® Communication Manager Hardware Description and Reference
Describes the hardware requirements for Avaya Aura® Communication Manager.
Avaya Aura® Communication Manager Survivability Options
Describes the system survivability options for Avaya Aura® Communication Manager.
Avaya Aura® Core Solution Description
Provides a high level description for the solution.
Maintenance and Troubleshooting
Avaya Aura® Communication Manager Reports
Describes the reports for Avaya Aura® Communication Manager.
Maintenance Procedures for Avaya Aura® Communication Manager,
Branch Gateways and Servers
Provides procedures to maintain Avaya servers and gateways.
Audience
Sales Engineers, Solution Architects Sales Engineers, Solution Architects
Sales Engineers, Solution Architects Sales Engineers, Solution Architects
Sales Engineers, Solution Architects
Sales Engineers, Solution Architects Sales Engineers, Solution Architects
Sales Engineers, Solution Architects, Implementation Engineers, Support Personnel Sales Engineers, Solution Architects, Implementation Engineers, Support Personnel
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Resources
Title Maintenance Commands for Avaya Aura® Communication Manager, Branch Gateways and Servers
Avaya Aura® Communication Manager Alarms, Events, and Logs Reference
Administration Administering Avaya Aura® Communication Manager
Administering Network Connectivity on Avaya Aura® Communication Manager
Avaya Aura® Communication Manager SNMP Administration and Reference
Administering Avaya Aura® Communication Manager Server Options
Avaya Aura® Communication Manager Data Privacy Guidelines
Implementation and Upgrading Deploying Avaya Aura® Communication Manager in Virtualized Environment
Deploying Avaya Aura® Communication Manager in Virtual Appliance
Deploying Avaya Aura® Communication Manager in Infrastructure as a Service Environment
Description Provides commands to monitor, test, and maintain Avaya servers and gateways.
Provides procedures to monitor, test, and maintain Avaya servers, and describes the denial events listed on the Events Report form.
Audience
Sales Engineers, Solution Architects, Implementation Engineers, Support Personnel
Sales Engineers, Solution Architects, Implementation Engineers, Support Personnel
Describes the procedures and screens for administering Communication Manager.
Describes the network connectivity for Communication Manager.
Describes SNMP administration for Communication Manager.
Describes server options for Communication Manager.
Describes how to administer Communication Manager to fulfill Data Privacy requirements.
Sales Engineers, Implementation Engineers, Support Personnel
Sales Engineers, Implementation Engineers, Support Personnel
Sales Engineers, Implementation Engineers, Support Personnel
Sales Engineers, Implementation Engineers, Support Personnel
Sales Engineers, Implementation Engineers, Support Personnel
Describes the implementation instructions while deploying Communication Manager on VMware and Kernel-based Virtual Machine (KVM).
Implementation Engineers, Support Personnel, Solution Architects
Describes the implementation instructions while deploying Communication Manager on Appliance Virtualization Platform.
Implementation Engineers, Support Personnel, Solution Architects
Describes the implementation instructions while deploying Communication Manager on Amazon Web Services, Microsoft Azure, Google Cloud Platform.
Implementation Engineers, Support Personnel, Solution Architects
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Communication Manager documentation
Title Deploying Avaya Aura® Communication Manager in Software-Only Environment
Upgrading Avaya Aura® Communication Manager
Understanding Avaya Aura® Communication Manager Feature Description and Implementation Avaya Aura® Communication Manager Screen Reference
Avaya Aura® Communication Manager Special Application Features
Description Describes the implementation instructions while deploying Communication Manager on a software-only environment.
Describes instructions while upgrading Communication Manager.
Audience
Implementation Engineers, Support Personnel, Solution Architects
Implementation Engineers, Support Personnel, Solution Architects
Describes the features that you can administer using Communication Manager.
Describes the screens that you can administer using Communication Manager.
Describes the special features that are requested by specific customers for their specific requirement.
Sales Engineers, Solution Architects, Support Personnel
Sales Engineers, Solution Architects, Support Personnel
Sales Engineers, Solution Architects, Avaya Business Partners, Support Personnel
Finding documents on the Avaya Support website
Procedure
1. Go to https://support.avaya.com.
2. At the top of the screen, type your username and password and click Login.
3. Click Support by Product > Documents.
4. In Enter your Product Here, type the product name and then select the product from the list.
5. In Choose Release, select the appropriate release number.
The Choose Release field is not available if there is only one release for the product.
6. In the Content Type filter, click a document type, or click Select All to see a list of all available documents.
For example, for user guides, click User Guides in the Content Type filter. The list only displays the documents for the selected category.
7. Click Enter.
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Resources
Accessing the port matrix document
Procedure 1. Go to https://support.avaya.com. 2. Log on to the Avaya website with a valid Avaya user ID and password. 3. On the Avaya Support page, click Support by Product > Documents. 4. In Enter Your Product Here, type the product name, and then select the product from the list of suggested product names. 5. In Choose Release, select the required release number. 6. In the Content Type filter, select one or both the following categories: · Application & Technical Notes · Design, Development & System Mgt The list displays the product-specific Port Matrix document. 7. Click Enter.
Avaya Documentation Center navigation
The latest customer documentation for some programs is now available on the Avaya Documentation Center website at https://documentation.avaya.com.
Important: For documents that are not available on Avaya Documentation Center, click More Sites > Support on the top menu to open https://support.avaya.com. Using the Avaya Documentation Center, you can: · Search for content by doing one of the following:
- Click Filters to select a product and then type key words in Search. - From Products & Solutions, select a solution category and product, and then select the
appropriate document from the list. · Sort documents on the search results page. · Click Languages ( ) to change the display language and view localized documents. · Publish a PDF of the current section in a document, the section and its subsections, or the
entire document. · Add content to your collection by using My Docs ( ).
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Training
Navigate to the Manage Content > My Docs menu, and do any of the following: - Create, rename, and delete a collection. - Add topics from various documents to a collection. - Save a PDF of selected content in a collection and download it to your computer. - Share content in a collection with others through email. - Receive collection that others have shared with you. · Add yourself as a watcher using the Watch icon ( ). Navigate to the Manage Content > Watchlist menu, and do the following: - Enable Include in email notification to receive email alerts. - Unwatch selected content, all content in a document, or all content on the Watch list page. As a watcher, you are notified when content is updated or deleted from a document, or the document is removed from the website. · Share a section on social media platforms, such as Facebook, LinkedIn, and Twitter. · Send feedback on a section and rate the content. Note: Some functionality is only available when you log on to the website. The available functionality depends on the role with which you are logged in.
Training
The following courses are available on the Avaya Learning website at www.avaya-learning.com. After logging into the website, enter the course code or the course title in the Search field and click Go to search for the course.
Course code 20970W 20980W 71200V 72200V 20130V 21450V
Course title Introducing Avaya Device Adapter What's New with Avaya Aura® Release 8.1 Integrating Avaya Aura® Core Components Supporting Avaya Aura® Core Components Administering Avaya Aura® System Manager Release 8.1 Administering Avaya Aura® Communication Manager Release 8.1
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Resources
Viewing Avaya Mentor videos
Avaya Mentor videos provide technical content on how to install, configure, and troubleshoot Avaya products. About this task Videos are available on the Avaya Support website, listed under the video document type, and on the Avaya-run channel on YouTube.
· To find videos on the Avaya Support website, go to https://support.avaya.com/ and do one of the following: - In Search, type Avaya Mentor Videos, click Clear All and select Video in the Content Type. - In Search, type the product name. On the Search Results page, click Clear All and select Video in the Content Type. The Video content type is displayed only when videos are available for that product. In the right pane, the page displays a list of available videos.
· To find the Avaya Mentor videos on YouTube, go to www.youtube.com/AvayaMentor and do one of the following: - Enter a key word or key words in the Search Channel to search for a specific product or topic. - Scroll down Playlists, and click a topic name to see the list of videos available for the topic. For example, Contact Centers. Note: Videos are not available for all products.
Support
Go to the Avaya Support website at https://support.avaya.com for the most up-to-date documentation, product notices, and knowledge articles. You can also search for release notes, downloads, and resolutions to issues. Use the online service request system to create a service request. Chat with live agents to get answers to questions, or request an agent to connect you to a support team if an issue requires additional expertise.
Using the Avaya InSite Knowledge Base
The Avaya InSite Knowledge Base is a web-based search engine that provides: · Up-to-date troubleshooting procedures and technical tips
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Support
· Information about service packs · Access to customer and technical documentation · Information about training and certification programs · Links to other pertinent information If you are an authorized Avaya Partner or a current Avaya customer with a support contract, you can access the Knowledge Base without extra cost. You must have a login account and a valid Sold-To number. Use the Avaya InSite Knowledge Base for any potential solutions to problems. 1. Go to http://www.avaya.com/support. 2. Log on to the Avaya website with a valid Avaya user ID and password.
The system displays the Avaya Support page. 3. Click Support by Product > Product-specific Support. 4. In Enter Product Name, enter the product, and press Enter. 5. Select the product from the list, and select a release. 6. Click the Technical Solutions tab to see articles. 7. Select relevant articles.
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Appendix A: PCN and PSN notifications
PCN and PSN notifications
Avaya issues a product-change notice (PCN) for any software update. For example, a PCN must accompany a service pack or an update that must be applied universally. Avaya issues a productsupport notice (PSN) when there is no update, service pack, or release fix, but the business unit or Avaya Services need to alert Avaya Direct, Business Partners, and customers of a problem or a change in a product. A PSN can also be used to provide a work around for a known problem, steps to recover logs, or steps to recover software. Both these notices alert you to important issues that directly impact Avaya products.
Viewing PCNs and PSNs
About this task To view PCNs and PSNs, perform the following steps: Procedure
1. Go to the Avaya Support website at https://support.avaya.com. If the Avaya Support website displays the login page, enter your SSO login credentials.
2. On the top of the page, click DOCUMENTS. 3. On the Documents page, in the Enter Your Product Here field, type the name of the
product. 4. In the Choose Release field, select the specific release from the drop-down list. 5. Select the appropriate filters as per your search requirement.
For example, if you select Product Support Notices, the system displays only PSNs in the documents list. You can apply multiple filters to search for the required documents.
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Signing up for PCNs and PSNs
Signing up for PCNs and PSNs
About this task Manually viewing PCNs and PSNs is helpful, but you can also sign up for receiving notifications of new PCNs and PSNs. Signing up for notifications alerts you to specific issues you must be aware of. These notifications also alert you when new product documentation, new product patches, or new services packs are available. The Avaya Notifications process manages this proactive notification system. To sign up for notifications: Procedure
1. Go to http://support.avaya.com and search for "Avaya Support Web Tips and Troubleshooting: E-Notifications Management". Under the Results section, click Avaya Support Web Tips and Troubleshooting: ENotifications Management.
2. Set up e-notifications. For detailed information, see the How to set up your E-Notifications procedure.
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Index
Special Characters
;Ethernet Options; ..............................................................449 (Calls Warning) Port ...........................................................390 (Starting) Port .................................................................... 663 (Time Warning) Port ...........................................................397 % of Gateways in Network Region with Hyperactive Registration Alarms ........................................................... 481
Numerics
011000 .............................................................................678 1st/2nd/3rd Skill
Vector Directory Number ............................................ 991 1-Step Clearing ..................................................................785 2-Party Loss Plan/Tone Loss Plan .....................................574 3PCC H.323 Override SIP Station Activation/Deactivation ............................................................................................ 268 440Hz PBX-dial Tone .........................................................896 440Hz Secondary-dial Tone ...............................................896 6400/8400/2420J LINE APPEARANCE LED SETTINGS . 358 7103A Button Assignment
Abbreviated Dialing ...................................................... 34 7405ND Numeric Terminal Display ....................................304 7434ND ..............................................................................304 802.1p ................................................................................494 802.1P/Q Parameters ........................................................ 463
A
A/D Grp/Sys List Dialing Start at 01 ...................................828 AACC Conference Modes
Feature Access Codes ............................................... 272 AAR/ARS Access .............................................................. 305 AAR/ARS Access Code .............................................158, 208 AAR/ARS Dial Tone Required ........................................... 284 AAR/ARS Internal Call Prefix .............................................214 AAR/ARS Internal Call Total Length .................................. 214 AAR/ARS Partitioning ........................................................ 127 AAR and ARS Digit Conversion Table
ANI Reqd ...................................................................... 24 AAR Digit Analysis Table ..................................................... 19 AAR Digit Conversion Table ................................................ 24 AAS ....................................................................................399
Agent Login ID ..............................................................45 Abandoned Call Search .....................................................939 Abbreviated Dial ................................................................ 254 abbreviated dialing .............................................................164 Abbreviated Dialing ............................................................290
Enhanced list ................................................................ 27 Group list ...................................................................... 27 Personal list .................................................................. 27 System list .................................................................... 27
ABBREVIATED DIALING .....................................................72 Abbreviated Dialing Enhanced List ....................................828 Abbreviated Dialing List 1, List 2, List 3 .............................781 Abbreviated Dialing List1 Access Code .............................253 Abbreviated Dialing List2 Access Code .............................254 Abbreviated Dialing List3 Access Code .............................254 abbreviated dialing lists ..................................................... 164 Abbreviated Dial Programming by Assigned Lists .............284 Abort Conference ...............................................................312 Abort Transfer .................................................................... 312 Absorption Treatment Assignment .....................................220 Absorption Treatment Information ..................................... 220 AC ........................................................................................40
Authorization Code COR Mapping ............................... 81 ACA Assignment ........................................................523, 939 ACA Long Holding Time Originating Extension ................. 284 ACA Referral Calls .............................................................284 ACA Referral Destination ...................................................285 ACA Remote PBX Identification ........................................ 285 ACA Short Holding Time Originating Extension .................285 Acceptable Service Level (sec) ......................................... 991 Accept Redirect to Blank User Destination ........................976 Access Code ......................................................................777 Access Code 2 ...................................................................254 access endpoint ...................................................................36 Access Endpoint .................................................................. 36
Communication Type ....................................................36 Extension ......................................................................37 Name ............................................................................ 37 TN .................................................................38, 516, 916 Width ............................................................................ 38 accessing port matrix .......................................................1016 Access Security Gateway (ASG) ....................................... 828 Access to MCT .................................................................. 121 ACD ........................................................................... 389, 842 ACD Login Identification Length ........................................ 339 Activate Answer Detection (Preserves SBA) On Final CCRON Cvg Point ............................................................. 814 Activation Code ..................................................................282 Active Coverage Path ............................................................175 Active Station Ringing ........................................................764 Act Time .............................................................542, 886, 888 Actual % .............................................................................654 ACW Agent Considered Idle Agent Login ID ..............................................................45 ACW Agents Considered Idle ............................................ 335 ACW Forced Logout Reason Code ................................... 347 Add/Remove Agent Skills .................................................. 122 Add Agent Skill Access Code ............................................ 273 add announcement .............................................................. 57 add attendant .......................................................................64 add audio-group ...................................................................78
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Index
Address Digits Include End-of-Digits Signal ...................... 609 Ad hoc Video Conferencing ............................................... 145 Adjunct CTI Link
Hunt Group ................................................................. 399 Adjunct Supervision ........................................................... 767 Administered Connection .....................................................40 Administered Members (min/max) .....................418, 546, 973 Administer secondary ip server interface board ................ 491 Administer Timers ...................................................... 516, 920 Administrable Timers ................................................. 539, 954 AE Services Server ............................................................502 After Call Work Access Code .............................................273 After Xfer or Held Call Drops ............................................. 996 AGENT AND CALL SELECTION ...................................... 335 Agent Login ID ..................................................................... 45 Agent Login ID page 1 ......................................................... 45 Agent Login ID page 2 ......................................................... 52 Agent States ...................................................................... 849 AGL ....................................................................................470 alarms
hyperactive registration .............................................. 481 alarms and detection ......................................................... 481 Alarm Threshold
Administered Connection ............................................. 43 Alarm Type
Administered Connection ............................................. 43 Alarm When PRI Endpoint Detached ................................ 238 Alert Incoming SIP Crisis Calls .......................................... 700
Signaling Group ..........................................................700 Alerting (sec) ......................................................................165 Alert Pager .........................................................................180 ALERT PAGER .................................................................. 180 ALERT STATION ............................................................... 179 Alias Set Type
Alias Station ..................................................................55 Alias Station .........................................................................55 Allocation Plan Number ..................................................... 543 Allow ANI Restriction on AAR/ARS ................................... 305 Allow ATMS Busyout, Error Logging and Alarming ............969 Allow Bridge DM Answer
Feature-related system parameters ........................... 361 Allow Conference via Flash ............................................... 324 Allow Crisis Alert Across Tenants ...................................... 324 Allow Direct-IP Multimedia .................................................440 allowed with exclusion
service observing ........................................................333 single step conference (SSC) .....................................333 Allow H.248 Gateways .......................................................449 Allow H.323 Endpoints .......................................................449 Allow Night Service Calls To Forward ................................158 Allow Ringer-off with Auto-Answer .....................................341 Allow SIP URI Conversion ................................................. 468 Allow Trunk Seizure On Busy Indicator ............................. 135 Allow VDN Override ...........................................................991 Alpha-name Alias Station ..................................................................56 Alphanumeric Dialing Table ................................................. 56
Alternate FRL Station ........................................................ 158 Alternate Route Timer ........................................................700 Always Use
Attendant Console ........................................................ 69 Station ........................................................................ 765 Always Use re-INVITE for Display Updates Trunk Group ................................................................977 AMD Treatment ..................................................................729 Analog Busy Auto Callback ............................................... 314 Analog Digital .....................................................................598 Analog Gain ....................................................................... 902 Analog Line Transmission Location Parameters .................................................. 568 Analog Loss Group .................................................... 644, 920 Analog Ringing Cadence Location Parameters .................................................. 568 Analog Trunk Incoming Call ID .......................................... 828 ANI/II-Digits ..........................................................................92 ANI Available ..................................................................... 612 ANI Not Available ...............................................................612 ANI Prefix ...........................................................................601 ANI Reqd AAR and ARS Digit Analysis Table ...............................19 ANI Source for Forwarded & Covered Calls ...................... 609 Annc Name Announcements Audio Sources ................................... 58 Annc Type Announcements Audio Sources ................................... 58 Announcement ...................................................................865 Announcement Access Code ............................................ 254 Announcement Extension ..................................................693 Announcement Ports ......................................................... 381 Announcements .........................................................314, 621 Announcements Audio Sources .......................................... 57 Announcement Server IP Address .................................... 479 Announcement Storage Path Name .................................. 479 Announcement Type ..........................................................381 Ans Delay .......................................................................... 973 Answer Back Access Code ................................................254 answer groups ................................................................... 171 Answer Send (msec) ......................................................... 955 Answer Supervision by Call Classifier ............................... 828 Answer Supervision Timeout ............................. 517, 644, 921 Answer Supervision Timeout (sec) .................................... 593 Answer Text ....................................................................... 197 APLT .................................................................................. 122 Appl ....................................................................................666 Appl. ...................................................................................726 application ..........................................................................790 Application Stations With Off-PBX Telephone Integration .............790 Apply Intercept Locally .......................................................328 Apply Local Ringback ........................................................ 523 Apply MCT Warning Tone .................................................. 302 Apply Ready Indication Tone To Which Parties In The Call ............................................................................................ 315 Apply Ringback for Auto Answer calls ............................. 1006
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Index
Apply Ringback upon Receipt of ........................................152 ARS ........................................................................... 255, 828 ARS/AAR Dialing without FAC ...........................................829 ARS/AAR Partitioning ........................................................ 829 ARS Digit Analysis Table ..................................................... 19 ARS Digit Conversion Table ................................................ 24 ARS FAC ........................................................................... 564 ARS Prefix 1 Required for 10-Digit NANP Calls ................ 565 ARS Toll Table ..................................................................... 63 ASAC Enabled ...................................................................621 ASAI ...........................................................................370, 540 ASAI Call Classification ..................................................... 337 ASAI Link Core Capabilities ...............................................829 ASAI Link Plus Capabilities ............................................... 829 ASAI Routing ....................................................................... 93 ASAI Uses Station Lock .....................................................130 ASCII mode ....................................................................... 379 As-needed Inactivity Time-out (min) .................................. 726 Assigned Member -- Ext ................................................... 787 Assigned Member -- Name ...............................................787 Assigned Members ............................................................ 432 Assigned VDN ................................................................... 244 Assignment .............................................................. 987, 1005 Assist Access Code ........................................................... 273 Assistance Call .................................................................. 167 Associated Signaling ......................................................... 701 Asynch. Transfer Mode (ATM) PNC .................................. 830 Asynch. Transfer Mode (ATM) Trunking ............................ 830 At End of Member List ....................................................... 418 ATMS ................................................................................. 830 ATMS Thresholds .............................................................. 967 ATM WAN Spare Processor .............................................. 830 Att. Call Waiting Indication ................................................. 764 Attd FAC ............................................................................ 565 Attendant Access Code ..................................................... 255 Attendant Console ............................................................... 64
display module buttion assignment .............................. 77 feature button assignment ............................................ 74 Softconsole IP Attendant .............................................. 69 VIS feature options ....................................................... 71 attendant console group parameters ................................. 158 Attendant Console X ............................................................64 Attendant Diversion Timing (sec) .......................................623 Attendant Group ................................................................ 869 Attendant Group Name ......................................................159 Attendant Lockout ..............................................................159 Attendant Originated Calls ......................................... 311, 872 Attendant Queue Announcement ...................................... 622 Attendant Time Out Flag ....................................................298 Attendant Tone;Auto Hold ..................................................305 Attendant Vectoring ............................................. 93, 830, 992 Attendant Vectoring VDN ...........................................159, 870 Audible Message Waiting .......................................... 766, 830 Audible Notification ............................................................ 364 Audio 802.1p Priority ......................................................... 463 Audio Codec ...................................................................... 435 Audio Group .........................................................................78
Audio PHB Value ............................................................... 462 Audio Resource Reservation Parameters ......................... 463 Audio Source Location
Audio Group ................................................................. 78 Audix
Agent Login ID ..............................................................46 AUDIX Mach ID ................................................................. 588 Audix-MSA Node Names .....................................................80 Audix Name ....................................................................... 765 AUDIX Name ......................................................411, 881, 996 Audix Name for Messaging
Agent Login ID ..............................................................46 Audix Names ....................................................................... 80 Auth Code ..................................................................504, 902 Authoritative Domain ......................................................... 459 Authorization Code Cancellation Symbol .......................... 298 Authorization Code COR Mapping ...................................... 81 Authorization Code Length ................................................ 298 AUTHORIZATION CODE PARAMETERS .........................298 Authorization Code Required .............................................674 Authorization Codes .......................................................... 831 Authorization Codes Enabled ............................................ 299 Auto ................................................................................... 494 Auto?
IP Interfaces ............................................................... 455 Auto-A/D ............................................................................ 781 Auto Abbreviated/Delayed Transition Interval (rings) ........ 285 Auto Alternate Routing .......................................................255 Auto Answer ...................................................................... 135
Agent Login ID ..............................................................46 Attendant Console ........................................................ 64 Station ........................................................................ 765 Auto-answer IP Failure AUX Reason Code .......................346 Auto Guard ........................................................................ 921 Auto Hold ........................................................................... 305 Auto-In Access Code ......................................................... 273 Auto Inspect on Send All Calls .......................................... 302 Automatic Callback ............................................................ 141 Automatic Callback Activation ........................................... 255 Automatic Callback -- No Answer Timeout Interval (rings) ............................................................................................ 286 Automatic Charge Display ................................................. 122 Automatic Circuit Assurance; Call Park Timeout;DID/Tie/ ISDN/SIP Treatment .......................................................... 286 Automatic Exclusion .......................................................... 141 Automatic Exclusion by COS .............................................353 Automatic Exclusion Coverage/Hold ................................. 353 Automatic Exclusion with Whisper Page ........................... 354 Automatic Moves ............................................................... 766 Automatic Selection of DID Numbers ........................ 382, 787 Automatic Wakeup .............................................................556 Automatic Wakeup Call Access Code ............................... 277 Auto Reserve Agents .........................................................335 Auto Restoration Administered Connection ............................................. 43 auto return ......................................................................... 856 day ..............................................................................857
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Index
auto return (continued) time .............................................................................857 time delay ................................................................... 858
Auto Return ........................................................................863 Auto Route Selection (ARS) Access Code 1 ..................... 255 Auto Select Any Idle Appearance ...................................... 766 Auto Start ...........................................................................305
Attendant Console ........................................................ 71 Auxiliary Board for Announcement .................................... 382 Aux Work ........................................................................... 672 Aux Work Access Code ..................................................... 274 AUX work mode ...................................................................47 Aux Work Reason Code Type ............................................345
Agent Login ID ..............................................................47 Available Agent Adjustments for BSR ................................330 Avaya Aura Media Server .................................................. 856 Avaya Aura MS .................................................................. 857 Avaya courses ................................................................. 1017 Avaya Media Server ...........................................................585 Avaya support website .....................................................1018
B
Backup Alerting ..................................................................159 Backup MFSS CCA-ID ...................................................... 365 Backup Servers in Priority Order ....................................... 468 Backward Cycle Timer (sec) .............................................. 601 Band .................................................................................. 683 Barge-In Tone .................................................................... 152 Barrier Code ...................................................................... 674 Barrier Code Length .......................................................... 675 Base Parameter Set .......................................................... 875 Base Tone Generator Set .................................................. 897 Basic
Call Vector .................................................................... 93 Basic Call Setup ................................................................ 848 Basic Mode Activation ....................................................... 278 Basic Supplementary Services .......................................... 848 basic vectoring ...................................................................845 Bcc
Attendant Console ........................................................ 72 BCC ................................................................... 190, 739, 903 BCC Value ......................................................................... 684 BCIE .................................................................................. 684 BCMS/VuStats Abandon Call Timer (seconds) ..................339 BCMS/VuStats LoginIDs ....................................................339 BCMS/VuStats Measurement Interval ............................... 339 BCMS/VuStats Service Level ............................................ 842 BCMS (Basic) .................................................................... 842 Bearer ................................................................................ 740 Bearer Capability Class ..................................................... 190 Bearer Capability Information Element .............................. 684 Beginning Station ...............................................................367 Begin Time .........................................................................890 Best Service Routing ................................................... 83, 845 Bit Rate .............................................................................. 224
ISDN Trunk Group ...................................................... 517
Bit Rate (continued) Trunk Group ................................................................921
Block CMS Move Agent Events .........................................186 Blocked Precedence Level ................................................ 622 Block Enhanced Conference Transfer Display
Class of Restriction .....................................................130 Block Hang-up by Logged-In Auto Answer Agents ............336 Block Progress Indicator ....................................................238 Block Sending Calling Party Location in INVITE ............... 978 Block Transfer Displays ..................................................... 131 Board ................................................................................. 489 Board Location ...................................................................424 Brazil Collect Call Blocking ................................................ 131 Break (msec) ..................................................................... 965 Bridged Appearance Origination Restriction ..............188, 777 Bridged Call Alerting .......................................................... 767 Bridged Calls ..................................................................... 793 Bridged Idle Line Preference ............................................. 767 Bridge ID ..........................................................................1008 Bridging Tone .....................................................................305 BRI LINK/MAINTENANCE PARAMETERS ....................... 199 BSR ..................................................................................... 93 BSR Application .................................................................997 BSR Available Agent Strategy
Vector Directory Number ............................................ 997 BSR Local Treatment .........................................................998 BSR Reply-best DISC Cause Value .................................. 523 BSR Tie Strategy ....................................................... 330, 998 Building
Station ........................................................................ 782 Build Refer-To URI of REFER From Contact For NCR
Trunk Group ................................................................978 bulletin board ....................................................................... 86 Bulletin Board text lines ....................................................... 86 Business Advocate ............................................................ 842 Busy
Coverage Path ............................................................175 Busy, Not Equipped ........................................................... 622 Busy/Reorder/Intercept Tone Timeout (Seconds) ..............351 Busy Auto Callback without Flash ..................................... 767 Busy Indicator for Call Parked on Analog Station Without Hardware? ......................................................................... 165 Busy Indicator lamp ........................................................... 165 Busy Out ............................................................................ 196 Busy Threshold ..........................................................505, 903 Busy Tone Disconnect ....................................................... 955 Busy Tone Disconnect for Analog Loop-start Trunks .........821 BUTTON ASSIGNMENTS ................................................. 784 button labels ...................................................................... 557 Button Type Customization Restriction ................................87 BW Management Option ................................................... 486 Bypass If IP Threshold Exceeded ......................................701
C
Cable ................................................................................. 782 Cadence Classification After Answer .................................859
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Index
Cadence Step ............................................................ 860, 897 call appearance buttons .....................................................557 Call Appearance Display Format ....................................... 769 Call Appearance Selection for Origination .........................152 Call Category for Vector ii-digits ........................................ 609 Call Center Elite .................................................................349 Call Center Release ...........................................................843 call clearing
tandem calls ............................................................... 979 Call Control 802.1p Priority ................................................463 Call Control PHB Value ......................................................463 Call Counts ........................................................................ 654 Call Coverage/Call Forwarding ..........................................807 Called Len ..........................................................................420 Called Number ...................................................................421 called number-only ............................................................ 981 Called Party Restriction ..................................................... 122 Caller ID calls .....................................................................286 caller ID format for Extension to Cellular telephones .........153 Caller ID for Service Link Call to H.323 1xC ......................922 Caller ID Message Waiting Indication ................................ 741 Caller ID on Call Waiting Delay Timer (msec) ....................324 Caller ID on Call Waiting Parameters ................................ 324 Caller's Emergency Service Identification (CESID) ........... 102 Call Forwarding Activation Busy/DA .................................. 255 Call Forwarding All Calls ....................................................142 Call Forwarding Busy/DA ...................................................142 Call Forwarding Enhanced ................................................ 145 Call Forwarding Enhanced Activation/Deactivation ........... 256 Call Forwarding Enhanced Status ..................................... 256 Call Forward Override ........................................................813 Call Handling Preference
Agent Login ID ..............................................................52 call identifiers ..................................................................... 558 Calling Number Style ......................................................... 153 Calling Number Verification ............................................... 153 Calling Party Number Conversion for Tandem Calls ........... 97 Calling Party Number to INTUITY AUDIX .......................... 411 Calling Party Restriction .................................................... 123 CALLING PERMISSION ....................................................138 Calling permissions ............................................................874 Call Limit
Stations With Off-PBX Telephone Integration .............793 Call Log Notify
Configuration Set ........................................................153 CALL MANAGEMENT SYSTEMS .....................................338 Call Origin .......................................................................... 998 Call Park Access Code ...................................................... 256 Call Park Timeout .............................................................. 296 Call Park Timeout Interval (minutes) ..................................286 Call Pickup .........................................................................363 Call Pickup Access Code ...................................................256 Call Pickup Alerting ............................................................363 Call Pickup for call to Coverage Answer Group .................363 Call Pickup on Intercom Calls ....................................295, 363 CALL PROCESSING OVERLOAD MITIGATION .............. 294 Call Rate .......................................................................... 1008
Callr-info Display Timer (sec) .............................................341 Calls Allowed ............................................................. 742, 794 Call Selection Measurement ..............................................336 Calls In Queue Warning .....................................................159 Calls Share IP Signaling Connection .................................701 Call Still Held ............................................................. 644, 922 Calls to Hunt Group -- Record .......................................... 105 Calls Used ......................................................................... 675 Calls Warning Extension ....................................................389 Calls Warning Threshold ....................................................390 Call Type ............................................................................210
Survivable ARS Analysis Table .................................. 796 Call Type AAR only
AAR and ARS Digit Analysis Table ...............................20 Call Type Analysis Behavior When Extension Not Assigned
Call Type Digit Analysis .............................................. 215 Call Type ARS only
AAR and ARS Digit Analysis Table ...............................20 Call Type Digit Analysis Table ..............................................89 Call-Type Ordering Within Priority Levels? ........................ 167 Call Vector ........................................................................... 91 Call Waiting Indication ....................................................... 768 Call Work Codes ................................................................ 843 CAMA Numbering format .....................................................69 CAMA Numbering Format ................................................. 102 Cama Outgoing Dial Guard (msec) ................................... 956 Cama Wink Start Time (msec) ...........................................956 Can Be a Service Observer ............................................... 123 Can Be Picked Up By Directed Call Pickup .......................124 Can Be Service Observed ................................................. 124 Cancel Do-Not-Disturb for Wakeup Calls .......................... 387 Can Change Coverage ...................................................... 124 Can Use Directed Call Pickup ........................................... 124 capacity profile max sessions ............................................ 586 capacity profile MPU estimate ........................................... 586 Carrier Medium .................................................................. 505 CAS ................................................................................... 159 CAS Back-Up Extension ....................................................160 CAS Branch ....................................................................... 831 CAS Main ...........................................................................831 CAS Remote Hold/Answer Hold-Unhold Access Code ..... 256 CA-TSC Request
Route Pattern ............................................................. 685 CBC Service Trunk Group Allocation Plan Assignment Schedule ............................................................................542 CBC Trunk Group Usage Allocation .................................. 543 CC ......................................................................................790
Stations With Off-PBX Telephone Integration .............791 CCA Enabled ..................................................................... 366 CCSA .................................................................................122 CDR ................................................................................... 579 CDR Account Code ........................................................... 257 CDR Account Code Length ............................................... 105 CDR Date Format .............................................................. 105 CDR FEAC ........................................................................ 892 CDR for Calls to EC500 Destination ..................................154 CDR for Origination ........................................................... 154
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Index
CDR Privacy ...................................................................... 768 CDR Reports ..................................................... 506, 641, 904 CDR retention .................................................................... 105 CDR system parameters ................................................... 104 Cell Phone Number ........................................................... 740 Cellular Voice Mail Detection
Configuration set ........................................................ 154 Centralized Attendant ........................................................ 848 Centralized Attendant Service (CAS) ........................ 158, 159 Centralized Automatic Message Accounting (CAMA) ....... 102 CESID ................................................................................102 CESID I Digits Sent ........................................................... 904 CF-CB Common ................................................................ 593 Chained Call Forwarding ................................................... 351 change aar analysis .............................................................19 change aar digit-conversion .................................................24 change abbreviated-dialing group ....................................... 29 change abbreviated-dialing personal ...................................31 change abbreviated-dialing system ..................................... 32 change administered-connection .........................................40 change agent-loginID
auto ...............................................................................45 change alias station ............................................................. 55 Change all board location translations from board ............ 427 change alphanumeric-dial-table ...........................................56 change ars analysis ............................................................. 19 change ars digit-conversion .................................................24 change ars toll ..................................................................... 63 change authorization-code .................................................. 81 change best-service-routing ................................................ 83 change bulletin-board .......................................................... 86 change button restriction ..................................................... 87 change calltype analysis ......................................................89 Change COR by FAC ........................................................ 831 Change Coverage Access Code ....................................... 257 Change Day (Start) ............................................................205 Change Day (Stop) ............................................................ 205 change media-server ......................................................... 585 Change Station Extension ..................................................117 change vector ...................................................................... 91 Channel Numbering ...........................................................224 Chan Port ...........................................................................723 Character Set for QSIG Name ...........................................535 Charge Advice ........................................................... 506, 517 Charge Conversion ............................................524, 645, 939 Charge Display Update Frequency (seconds) ...................325 Charge Type .............................................................. 645, 939 Check-In status function .................................................... 142 Check-Out status function ................................................. 142 chime code extensions ...................................................... 150 chime codes .......................................................................150 Chk Sum ............................................................................ 424 CINFO ..........................................................................93, 846 Circuit Pack Assignments .................................................. 597 Circuit Pack Location ......................................................... 597 Circuit Packs ......................................................................120 Circuit Type
Circuit Type (continued) Signaling Group ..........................................................702
City/Area ............................................................................ 566 class of restriction
calling permission ....................................................... 138 Service Observing Permission ................................... 138 Class of Restriction ....................................................121, 136 Class of service ................................................................. 140 Class of service: page 1 Automatic Callback .....................................................141 Automatic Exclusion ................................................... 141 Call Forwarding Busy/D Client Room ......................... 141 Console Permissions ..................................................141 class of service page 2 Ad hoc Video Conferencing ........................................ 145 Call Forwarding Enhanced ......................................... 145 Masking CPN/Name Override .................................... 145 Priority Ip Video .......................................................... 145 Clear Callr-info ...................................................................341 Clear-channel Mode .......................................................... 440 Clear VuStats Shift Data .................................................... 340 Client Room ....................................................................... 142 Client Room Class of Service .............................................. 65 Client Room Coverage Path Configuration ........................377 Clock Time Forced Logout Reason Code ..........................348 clustered ............................................................................ 702 Cluster ID ...........................................................................798 cluster Session Manager ................................................... 148 CMS appl mis Feature-related system parameters ........................... 338 Cntry/Peer Protocol ................................................... 548, 552 coach on single step conference (SSC) ............................ 333 Code .......................................................................... 546, 974 Code/Sfx ............................................................................ 450 Code Calling, COR ............................................................ 579 Code Calling, TAC ............................................................. 580 Code Calling, TN ............................................................... 580 Code Calling IDs ................................................................150 Code Calling Playing Cycles ..............................................579 Codec Set .................................................................. 436, 461 codec-set ........................................................................... 470 Code-Pattern Choice Assignments ....................................682 CODES .............................................................................. 682 Codeset to Send Display ................................................... 518 Codeset to Send National IEs ............................................518 Collect All Digits Before Seizure ........................................ 602 Collected Digits ..................................................................541 collection delete ........................................................................1016 edit name ..................................................................1016 generating PDF ........................................................ 1016 sharing content ......................................................... 1016 command parameters ........................................................790 Commercial Cost Avoidance (CCA) Related System Parameters ........................................................................ 365 Comm Type ....................................................................... 904 Communication Interface Processor Channels ................. 666
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Index
Communication Manager ...................................585, 692, 857 Community .................................................................798, 862 Community Size .................................................................799 Community String .............................................................. 483 Companding Mode ............................................................ 569 Computer Telephony Adjunct Links ................................... 831 Condition Code `T' for Redirected Calls .............................106 Conf/Trans On Primary Appearance ..................................769 Conference/Transfer .......................................................... 312 Conference Access Code ................................................ 1003 Conference Controller ......................................................1003 conference parties ............................................................. 306 Conference Tone ............................................................... 306 conference tone on service observing ............................... 333 Conference Type ............................................................. 1004 Configuration ..................................................................... 196 configuration set ................................................................ 790
location for routing incoming overlap calls ..................156 Configuration Set ............................................... 151, 742, 791 Configuration Set Description ............................................ 155 Confirmed Answer ............................................................. 155 Connect ............................................................................. 225
Signaling Group ..........................................................702 Connected Indication ......................................................... 197 Connected to ..................................................................... 190 Connected to CO ............................................................... 939 Connection Number
Administered Connection ............................................. 40 Connectivity Time .............................................................. 804 Connectivity Timer ............................................................. 501 CONNECT Reliable When Call Leaves ISDN ................... 518 Console parameters .......................................................... 158 Console Parameters
abbreviated dialing ..................................................... 164 common shared extensions ........................................165 incoming call reminders ..............................................165 timing .......................................................................... 166 Console parameters: page 3 ............................................. 167 Console Permissions ......................................................... 142 Console Type Attendant Console ........................................................ 64 Contact Closure Activation ................................................ 142 Contact Closure Close Code ............................................. 257 Contact Closure Open Code ..............................................258 Contact Closure Pulse Code ............................................. 258 content publishing PDF output .............................................. 1016 searching .................................................................. 1016 sharing ......................................................................1016 sort by last updated .................................................. 1016 watching for updates ................................................ 1016 Contiguous .................................................................545, 664 Continue Daily Until Completed .........................................367 Continuous Administered Connection ............................................. 41 Controlled Outward Restriction Intercept Treatment ..........295 Controlled Restrictions Configuration ................................ 377
Controlled Station to Station Restriction ............................ 296 Controlled Termination Restriction .....................................296 Controlled Toll Restriction Intercept Treatment ..................299 Controlled Toll Restriction Replaces .................................. 299 Controlling Adjunct
Hunt Group ................................................................. 400 Conv .................................................................................. 983
AAR and ARS Digit Conversion Table ..................25, 660 Converse Data Return Code ............................................. 276 Converse First Data Relay .................................................330 Converse Second Data Relay ........................................... 331 Converse Signaling Pause ................................................ 331 Converse Signaling Tone ...................................................331 Conversion to Full Public Number - Delete ........................467 Conversion to Full Public Number - Insert ......................... 467 Convert 180 to 183 for Early Media ................................... 978 Convert First Digit End-of-Dial To ...................................... 602 Convert ISDN Public Numbers to International for Display
............................................................................. 569 Location Parameters .................................................. 569 Copy ASAI UUI During Conference/Transfer .....................336 COR ...................................185, 507, 662, 675, 741, 905, 992 Access Endpoint ...........................................................37 Agent Login ID ..............................................................47 Announcements Audio Sources ................................... 59 Attendant Console .................................................. 65, 72 Authorization Code COR Mapping ............................... 81 console parameters .................................................... 160 Data Module ............................................................... 191 Group Paging Using Speakerphone ...........................371 Hunt Group ................................................................. 391 Terminating Extension Group ..................................... 881 COR/FRL check for Covered and Forwarded Calls ...........808 COR Description ................................................................124 Cord Length ....................................................................... 782 COR Number ..................................................................... 125 COR to Use for DPT .......................................................... 301 COS ................................................................... 191, 662, 675 Access Endpoint ...........................................................37 Attendant Console .................................................. 65, 72 console parameters .................................................... 161 MOC Control ...............................................................146 Station ........................................................................ 741 COS Group ................................................................ 142, 870 COS Name ........................................................................ 143 CO Type .............................................................................905 Count ................................................................................. 165 Country ...................................................................... 645, 905 Country code for CDR ....................................................... 569 country options table ..........................................................817 Country Protocol ................................................ 225, 703, 787 Coverage After Forwarding ........................................768, 813 Coverage Answer Group ................................................... 171 Coverage - Caller Response Interval (seconds) ................808 Coverage Module .............................................................. 787 Coverage Msg Retrieval .................................................... 768 Coverage Of Calls Redirected Off-Net Enabled ................ 815
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Index
coverage path .................................................................... 257 Coverage Path ...................................................173, 391, 642
Agent Login ID ..............................................................47 Terminating Extension Group ..................................... 881 Coverage Path 1 or Coverage Path 2 ................................740 Coverage Path Number ..................................................... 173 CPE LOOPBACK JACK OPTIONS ................................... 241 CPN, ANI for Dissociated Sets .......................................... 292 CPN/ANI/ICLID Parameters .............................................. 321 CPN/ANI/ICLID Replacement for Restricted Calls ............ 321 CPN/ANI/ICLID Replacement for Unavailable Calls ..........321 CPN Prefix ................................................................... 99, 637 CRC ................................................................................... 226 Create Universal Call ID (UCID) ........................................ 303 Crisis Alert Code ................................................................180 Crisis Alert System Parameters .........................................179 Crisis Alert to a Digital Pager .............................................180 Criteria for Logged Off/PSA/TTI Stations ........................... 811 Critical Reliable Bearer ...................................................... 450 CTI link ...............................................................................188 CTI Link ..................................................................... 184, 185 CTI Link Number ................................................................366 CTI Stations System Parameter Customer Options ........................850 Currency Symbol ............................................... 524, 646, 940 Customer Telephone Activation (CTA) Enabled .................292 Customizable Labels ......................................................... 741 Customize ..........................................................................575 Customize Parameters ...................................................... 875 Custom Selection of VIP DID Numbers ..................... 383, 788 Cut-Through .......................................................................922 Cvg Enabled for VDN Route-To Party ............................... 174 Cvg Of Calls Redirected Off-net ........................................ 831 CVG Path ...........................................................................887 Cyclical Hunt ......................................................................922
D
dadmin login ...................................................................... 833 Daily Wakeup .....................................................................383 Data Extension
Attendant Console ........................................ 72, 191, 785 Data Item ............................................................................114 Data Module .............................................................. 190, 191
Attendant Console ........................................................ 65 Station ........................................................................ 743 Data Option ........................................................................788 Data Origination Access Code ...........................................259 Data Privacy ...................................................................... 143 Data Privacy Access Code ................................................ 259 Data Restriction ......................................... 524, 642, 769, 940 Date Bulletin Board ............................................................... 86 Date (Start) ........................................................................ 205 Date (Stop) ........................................................................ 205 Date and Time ................................................................... 201 Date Format on Terminals ......................................... 325, 570
Day system parameters port networks .............................. 863
Daylight Saving Rule ............................................... 202, 1006 Daylight Saving Rules ........................................................204 Day of the Month ............................................................... 201 Day of the Week ................................................................ 202 DC ......................................................................................428 DCC Activation/Deactivation ..............................................146 D Channel ..........................................................................703 D-Channel ..........................................................................226 DCP/Analog Bearer Capability .......................... 548, 552, 704 DCP/ANALOG Bearer Capability .......................................226 DCP Terminal-parameters Plan ......................................... 570 DCS/QSIG Intw ..................................................................685 DCS (Basic) ....................................................................... 832 DCS Call Coverage ........................................................... 832 DCS to QSIG TSC Gateway ..............................................208 DCS with Rerouting ........................................................... 832 Deactivation Code ............................................................. 282 Debug Filter Values ............................................................497 Decimal Point .....................................................524, 646, 940 dedicated Voip channel licenses ........................................586 Default ANI ........................................................................ 602 Default Announcement Extension ..................................... 383 Default Call Appearance Display Format ...........................221 Default COR ...................................................................... 292 Default Coverage Path for Client Rooms ...........................378 DEFAULT DIALING ..............................................................73 Default Dialing Abbreviated Dialing Dial Code .................. 785 Default Multimedia Outgoing Trunk Parameter Selection ..290 Default Reason Code ........................................................ 672 Default Route Digit .............................................................623 Default RTCP Report Period (secs) ...................................480 Default Server IP Address ................................................. 480 Default Server Port ............................................................ 480 Default Service Domain ..................................................... 624 Definition for Rooms in State 1 through 6 ..........................387 Del ..................................................................................... 984
AAR Digit Conversion Table ................................. 25, 660 ARS Digit Conversion Table ................................. 25, 660 Incoming Call Handling Treatment ............................. 421 Delay feature related system parameters .............................329 Delay Call Setup When Accessed Via IGAR ..................... 923 Delay for USNI Calling Name for Analog Caller ID Phones (seconds) ........................................................................... 316 Delay Sending Release ..................................................... 302 Delete .................................................................................. 99 Call Type Digit Analysis Table ...................................... 89 Deluxe Paging and Call Park Timeout to Originator .......... 296 Deny Survivable ARS Analysis Table .................................. 796 Description .......................................................628, 988, 1005 Holiday Table .............................................................. 374 Service Hours Table ................................................... 698 Desktop Video Conferencing ....................................... 71, 776 Dest. Digits ........................................................................ 726
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Index
Dest # 1, 2, or 3 IP Address ...............................................496 Dest # IP address .............................................................. 484 Destination .........................................................................992
Administered Connection ............................................. 40 Destination Code Control ...........................................281, 657 Destination Node ....................................................... 489, 667 Destination Port ................................................................. 667 detection and alarms ......................................................... 481 Detect Slips ........................................................................549 DHCP .........................................................................492, 495 Dial Access ................................................................ 507, 906 Dial Code
Enhanced List .............................................28, 29, 31, 32 Personal List ...............................................28, 29, 31, 32 System List ................................................. 28, 29, 31, 32 DIAL CODE 7103A Button Assignment ............................................ 35 Dial Detection .................................................................... 923 Dial Echoing .......................................................................197 Dialed String .......................................................................211 AAR and ARS Digit Analysis Table .......21, 657, 797, 892 AAR and ARS Digit Conversion Table ..21, 657, 797, 892 Dialed String length (Min, Max) ........................................... 89 Dialed String Match Call Type Digit Analysis Table ...................................... 89 Dial Plan Analysis Table .................................................... 210 Dial Plan Parameters .........................................................214 Dial Plan Transparency in Survivable Mode ...................... 467 dial prefix ........................................................................... 790 Dial Prefix .................................................................. 743, 791 Dial Tone Validation Timer (sec) ........................................ 820 DID/Tie/ISDN/SIP Intercept Treatment .............................. 286 DID Busy Treatment .......................................................... 306 DID-LDN Only to LDN Night Ext ........................................ 161 DID to Attendant ................................................................ 168 DiffServ .............................................................................. 494 Diffserv/TOS Parameters ...........................................462, 466 Digit Absorption ................................................................. 220 Digit Absorption List ...........................................................906 Digital Loss Group ............................................. 519, 647, 923 Digital Loss Plan Modification ............................................832 digital PPM per country protocol ........................................235 Digit Handling (in/out) ........................................................ 519 Digits ..................................................................................923 Digits to Record for Outgoing Calls ................................... 106 Digit to Insert/Delete .......................................................... 383 Digit Treatment .................................................................. 924 Direct Agent Announcement Extension ............................. 330 Direct Agent Calling ........................................................... 125 direct agent calls ..................................................................48 Direct Agents Calls First Agent Login ID ..............................................................48 Direct Agent Skill Agent Login ID ..............................................................53 Direct Dial Access-Trunk ................................................... 590 Directed Call Pickup .................................................. 124, 363 Directed Call Pickup Access Code
Directed Call Pickup Access Code (continued) Feature Access Code page 2 ..................................... 259
Directed Group Call Pickup Access Code ......................... 259 Direct Inside Access .......................................................... 590 Direction .............................................................507, 594, 907 Direct IP-IP Audio Connections ......................................... 704
Attendant Console ................................ 69, 355, 361, 770 Directory Number ...............................................................554 Directory Search Sort Order .............................................. 820 Direct Trunk Group Select Button Assignments
Attendant Console ........................................................ 68 direct-WAN .........................................................................470 Disable call classifier for CCRON over ISDN trunks ..........815 Disable call classifier for CCRON over SIP trunks ............ 815 Disable Following a Security Violation ............................... 675 Disable Restarts ................................................................ 226 Disconnect Information in Place of FRL ............................ 106 Disconnect on No Answer by Call Type .............................820 Disconnect Sequence ........................................................198 Disconnect Signal Error
Trunk Group ................................................................956 Disconnect Supervision-In ......................................... 647, 924 Disconnect Supervision-Out ...................................... 519, 925 Disconnect Timing (msec) ......................................... 573, 925 Disconnect Type ................................................................ 925 Disp Client Redir
Attendant Console ........................................................ 65 Display Authorization Code ............................................... 299 Display between Locations with these Display Parameters ............................................................................................ 221 Display Caller ID ................................................................ 743 Display Calling Number for Room to Room Caller ID Calls ............................................................................................ 286 Display Cartridge ............................................................... 743 Display Character Set ........................................................821 Display Client Redirection ..................................................769 Display Connected Name/Number for ISDN DCS Calls ....316 Display Information With Bridged Call ............................... 358 Display Language .............................................................. 743
Attendant Console ........................................................ 66 Display Mode ..................................................................... 876 Display Module Button Assignments ................................... 77 Display Notification for a locked Station ............................ 351 Display Notification for Call Forward ..................................351 Display Notification for Do Not Disturb .............................. 351 Display Notification for Enhanced Call Forward ................ 351 Display Notification for Limit Number of Concurrent Calls .351 Display Notification for Posted Messages ......................... 352 Display Notification for Send All Calls ................................352 Display Parameters ........................................................... 221 Display PBX data on telephone ......................................... 265 Display Room Information in Call Display ..........................384 Display Text ....................................................................... 322 Display VDN for Route-To DAC ......................................... 999 Disp Parm .......................................................................... 565 Distinctive Audible Alert
Station ........................................................................ 770
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Index
Distinctive Audible Alerting .................................311, 872, 873 Diversion by Reroute ......................................................... 535 Diverted Party Identification ...............................................808 DLI Voltage Level ...............................................................876 DMI-BOS ........................................................................... 227 DND/SAC/Go to Cover
All ................................................................................176 Coverage Path ............................................................176 Don't Answer .............................................................. 176 documentation Communication Manager ......................................... 1013 documentation center ...................................................... 1016 finding content .......................................................... 1016 navigation ................................................................. 1016 documentation portal ....................................................... 1016 finding content .......................................................... 1016 navigation ................................................................. 1016 document changes .............................................................. 16 Domain .............................................................................. 251 Do Not Disturb Restriction ................................................. 296 Do Not Send Group B Signals to CO .................................610 Do not share port with unnamed registration .....................778 Download Flag ...........................................................483, 484 Download Set Type ............................................................367 Download Station Type ..............................................367, 885 Drop Parking User From the Call After Timeout ................ 297 Drop Treatment ..................................................................926 DS1 Circuit Pack ................................................................224 DS1 circuit pack: page 2 ....................................................240 DS1 Echo Cancellation ......................................525, 832, 941 DS1 MSP ........................................................................... 832 DSN Number ..................................................................... 788 DSN Term .................................................................. 525, 941 DST ....................................................................................567 dst rgn ................................................................................470 DTMF Duration On ............................................................ 591 DTMF Duration - Tone (msec) ........................................... 180 DTMF Feedback Signals For VRU .................................... 843 DTMF over IP .................................................................... 704 DTMF Tone Feedback ....................................................... 307 Dual Wakeup ..................................................................... 384 Duplex ........................................................................494, 594 IP Interfaces ............................................................... 455 ISDN Trunk Group ...................................................... 520 Trunk Group ................................................................926 Duplicate Station ................................................................243 Duplicate Vector .................................................................244 Duration Administered Connection ............................................. 42 Duration (msec) ................................................................. 897 Duration Maximum .............................................................860 Duration Minimum ..............................................................861 Duration of Call Timer Display ........................................... 354 DVC ............................................................................. 71, 776 Dynamic Advocate .............................................................843 Dynamic Percentage Adjustment ...................................... 400 Dynamic Queue Position ................................................... 400
Dynamic Threshold Adjustment .................................401, 843 Dyn CAC ............................................................................471
E
EAS ................................................................................45, 94 EAS-PHD .....................................................................54, 843 EC500delay deactivation state .......................................... 155 EC500 Self Administration Access Code ...........................259 EC500 state ....................................................................... 770 EC Configuration ............................................................... 238 EC Direction
DS1 Circuit Pack ........................................................ 239 Echo Cancellation ..............................................................239 Echo Digits Dialed ............................................................... 71 echo direction .................................................................... 239 ECM ...................................................................................440 Edit Dialing on 96xx H.323 Terminals ................................ 325 Emergency Access ............................................................ 168 Emergency Access Redirection Extension ................ 297, 301 Emergency Access to Attendant ........................................833 Emergency Access To Attendant Access Code .................259 Emergency Location Ext ....................................................770
Attendant Console ........................................................ 69 emergency location extension ............................................117 Emergency Location Extension ..................................118, 433 EMU Inactivity Interval for Deactivation ............................. 294 EMU Login Allowed ........................................................... 745 Enable ................................................................................667
Administered Connection ............................................. 40 Enable Busy Tone Disconnect for Analog Loop-start Trunks ............................................................................................ 821 Enable CDR Storage on Disk ............................................ 106 Enable Command Logging ................................................ 577 Enabled ......................................................................499, 502
Signaling Group ..........................................................726 Survivable Processor ..................................................802 Enable Detection and Alarms ............................................ 481 Enable Dial Plan Transparency in Survivable Mode ..........301 Enable Enbloc Dialing without ARS FAC ...........................323 Enable Ethernet Interface .......................................... 450, 455 Enable Failover Event Package Subscription ....................724 Enable File Transfer ...........................................................246 Enable Inter-Gateway Alternate Routing ........................... 301 Enable Layer 3 Test ........................................................... 705 Enable on Survivable Processors (ESS and LSP) ............ 723 Enable on the Main Processor(s) ...................................... 723 Enable Operation of IPSI Duplication ................................ 852 Enable Operation of PNC Duplication ............................... 852 Enable PE for H.248 Gateways ......................................... 799 Enable PE for H.323 Endpoints ......................................... 799 Enable QoS ....................................................................... 491 Enable Q-SIP .....................................................................979 Enable Reachability Station Domain ........................................................... 785 Enable Session ..................................................................247 Enable SIP Agent Reachability? ........................................349
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Index
Enable Syslog ....................................................................496 Enable Voice/Network Stats .............................................. 478 Enable VoIP/Network Thresholds ...................................... 457 Enable `dadmin' Login ....................................................... 833 encrypted SRTCP .............................................................. 437 Encryption ..........................................................................491 End Day ............................................................................. 374 End Hour ............................................................................375 Ending Station ................................................................... 368 End Min ..............................................................................375 End Month ......................................................................... 375 End OCM After Answer (msec) ......................................... 570 End of OCM Intercept Extension ....................................... 571 Endpoint .............................................................................300 Endpt ID .....................................................................554, 745 Endpt Init ............................................................199, 554, 744 End Time ............................................................................890 End to end signaling .................................................. 539, 965 End-to-End Signaling .........................................................926 End-to-End total loss (dB) in a n-party conference ............575 Enforce PNT-to-PNT Restrictions ...................................... 329 Enforce SIPS URI for SRTP .............................................. 705 Enhanced Abbreviated Dialing .......................................... 557 Enhanced Abbreviated Dial Length ................................... 290 Enhanced Attendant Notification ....................................... 164 Enhanced Call Pickup Alerting .......................................... 363 Enhanced Call Pickup Delay Timer (sec.) Display .............364 Enhanced Callr-Info display for 1-line display phones .......778 Enhanced Conferencing
System parameters customer options ........................ 833 Enhanced EC500 ...............................................................833 Enhanced EC500 Activation .............................................. 260 Enhanced EC500 Deactivation ..........................................260 Enhanced List
Abbreviated Dialing ...................................................... 27 Enhanced Mode Activation ................................................ 279 Enhanced PSA Location/Display Information Enabled ......293 Enterprise Mobility User Activation .................................... 260 Enterprise Mobility User Deactivation ................................260 Enterprise Survivable Server ............................................. 833 Enterprise Wide Licensing ................................................. 834 EPN ................................................................................... 695 EPSCS ...............................................................................122 Erase 24xx/14xx/94xx User Data Upon:
Dissociate or unmerge this phone .............................. 135 EMU login or logoff at this phone ................................135 ESF DATA LINK OPTIONS ................................................241 ESS Administration ............................................................833 Established ........................................................................ 726 ETA Node Number .............................................................215 ETA Routing Pattern .......................................................... 215 Ethernet Link ..............................................................450, 455 ETSI CCBS ........................................................................549 ETSI CCBS Support .......................................................... 706 Event Minimization .............................................................186 Every User Responds ........................................................179 Expand ISDN Numbers to International for 1XCES ...........362
Expansion Module ............................................................. 744 Expected Call Handling Time (sec) ....................................401 Expected Digits ..................................................................927 Expert Agent Selection ........................................................ 45 Expert Agent Selection (EAS) ............................................843 Expert Agent Selection (EAS) Enabled ............................. 330 Expert Agent Selection-Preference Handling Distribution ... 54 Expiration Date .................................................................. 676 Ext ......................................................................150, 651, 652
Attendant Console ........................................................ 74 Coverage Answer Group ............................................ 172 Group Paging Using Speakerphone ...........................371 Hunt Group ................................................................. 419 Intercom Group ...........................................................428 Listed Directory Numbers ........................................... 563 Terminating Extension Group ..................................... 883 Ext Alert (TAAS) Extension ........................................ 161, 871 Ext Alert Port (TAAS) ................................................. 161, 870 Ext and Name .................................................................... 195 Ext Code .................................................................... 633, 637 CAMA Numbering Format .......................................... 102 Extended Call Fwd Activate All ..........................................260 Extended Call Fwd Activate Busy D/A ...............................261 Extended Cvg/Fwd Admin ................................................. 833 Extended Forwarding All ....................................................143 Extended Forwarding B/DA ............................................... 143 Extended Group Call Pickup ......................................297, 364 Extended Group Call Pickup Access Code ....................... 261 Extended Group Number ...........................................248, 652 Extended Loop Range ....................................................... 927 Extended Pick-Up Group ................................................... 248 Extension ........................................................... 250, 662, 992 Announcements Audio Sources ................................... 59 Attendant Console ........................................................ 66 CTI Link ...................................................................... 185 Intra-Switch CDR ........................................................432 Station ........................................................................ 744 EXTENSION DISPLAY FORMATS ............................217, 221 Extension only label for Team button on 96xx H.323 terminals ............................................................................................ 322 Extensions Administered to have an MCT-Control Button .249 extensions for chime codes ............................................... 150 Extensions to Call Which Activate Features by Name .......250 Extension to Cellular call security ...................................... 152 Extension to Receive Failed Wakeup LWC Messages ...... 384 External Coverage ............................................................. 590 External Coverage Treatment for Transferred Incoming Trunk Calls ................................................................................... 811 External Device Alarm Admin ............................................ 834 External Ringing for Calls with Trunks ............................... 313 Ext Len .......................................................................634, 638 CAMA Numbering Format .......................................... 102
F
Facilities Restriction Level ................................................. 125 Facility Access Trunk Test ................................................. 125
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Avaya Aura® Communication Manager Screen Reference
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Index
Facility Busy Indicators ...................................................... 270 Facility Coding ................................................................... 630 Facility Test Calls Access Code .........................................261 Facility Type .......................................................................630 Failover/failback Signaling-group Pair ............................... 724 Failover Group Domain Mapping .......................................251 Far-end CSU Address ....................................................... 241 Far-end Domain .................................................................706 Far-end Listen Port ............................................................ 707 Far-end Network Region ....................................................707 Far-end Node Name
Signaling Group ..........................................................707 Far End Resource Info .....................................................1008 Far End Test Line No. ........................................................ 508 Far-End Test No .................................................................968 Fast Connect on Origination
Configuration Set ........................................................156 FAX Mode
IP codec set ................................................................441 FEAC ................................................................................. 125 Feature Access Code ........................................................ 253 Feature Access Code (FAC):Refresh Terminal Parameters Access Code ......................................................................267 Feature Access Code page 2
Directed Call Pickup Access Code ............................. 259 FEATURE BUTTON ASSIGNMENTS ................................. 74 Feature Invocation by In-Call DTMF Code
Configuration Set ........................................................156 Feature Module ..................................................................788 Feature Plus Ext ................................................................ 317 Feature-related system parameters ...................................283 Filename in Memory .......................................................... 884 File Size ............................................................................. 884 File Status ..........................................................................884 File to Retrieve ...................................................................884 finding content on documentation center .........................1016 finding port matrix ............................................................ 1016 Firmware Station Download ...............................................367 First Announcement Delay (sec) ....................................... 412 First Announcement Extension ..........................................412 Five Port Networks Max Per MCC .....................................834 Fixed .................................................................................. 543 Fixed TEI ................................................................... 200, 745 Flash .......................................................................... 170, 624 Flash Access Code ....................................................261, 281 Flashhook Interval ............................................................. 573 Flash Length (msec) .......................................................... 957 Flash Override ........................................................... 170, 624 Flash Override Access Code ............................................. 281 Flexible Billing ....................................................................834 Floor
Station ........................................................................ 782 Forced Agent Aux Work by Location Reason Code .......... 348 Forced Agent Aux Work by Skill Reason Code ................. 348 Forced Agent Logout by Location Reason Code ...............348 Forced Agent Logout by Skill Reason Code ......................348 Forced Agent Logout Time
Forced Agent Logout Time (continued) Agent Login ID ..............................................................48
Forced Entry of Account Codes .................................125, 834 Forced Entry of Stroke Counts or Call Work Codes .......... 408 Forced SIP Agent Logout for Unreachable Reason Code .348 Force Entry of Acct Code for Calls Marked on Toll Analysis Form .................................................................................. 106 Force Phones and Gateways to Active LSPs .................... 480 Format ....................................................................... 525, 927 Forward Cycle Timer (sec) .................................................603 Forward Disconnect Timer (msec) .....................................573 Forwarded Destination .......................................................779 Forward PMS Message to INTUITY Lodging .................... 378 Frames Per Pkt ..................................................................437 Framing Mode ....................................................................227 Frequency ..........................................................................966 FRL ............................................................................ 125, 685 FROM / TO ........................................................................ 575 From IP Address ................................................................433 Fully Restricted Service ..................................................... 126
G
G3V4 Adv Route ..................................................................94 G3V4 Enhanced .................................................................. 94 G3 Version ......................................................................... 823 Gatekeeper Priority ............................................................451 Gateway .............................................................489, 492, 495 Gateway Node Name ................................................ 451, 456 Glare .................................................................................. 957 Glare Handling ...................................................................941 Global Call Classification ................................................... 834 Global Classifier Adjustment (dB) ......................................860 Group
Attendant Console ........................................................ 66 Group Controlled Restriction ............................................. 126 Group Control Restrict Activation/Deactivation ..................262 Group Extension ................................................ 371, 391, 881
Hunt Group ................................................................. 418 Group II Called Party Category ..........................................603 Group II Category For MFC ............................................... 126 Group List
Abbreviated Dialing ...................................................... 29 Group List ..................................................................... 30 Group Member Assignments ............................. 546, 652, 973 Group Name .............................................. 371, 508, 642, 907 Audio Group ................................................................. 79 Coverage Answer Group ............................................ 172 Hunt Group ................................................................. 391 Group Number ................... 371, 508, 594, 642, 653, 882, 907 Coverage Answer Group ............................................ 172 Hunt Group ......................................................... 392, 418 Intercom Group ...........................................................428 Signaling Group ..........................................................708 Group Page Zip Tone Timer (secs) ................................................. 372 Group Paging Using Speakerphone .................................. 370
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Avaya Aura® Communication Manager Screen Reference
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Index
Group Paging Using Speakerphone (continued) alert .............................................................................370
Group Port Announcements Audio Sources ................................... 59
Group Timeout (secs) ........................................................ 372 Group Type ................................ 508, 520, 594, 643, 907, 928
Hunt Group ......................................................... 392, 418 Signaling Group ..........................................................708 Grp No ............................................................................... 685 Gtwy to ...............................................................................667
H
H.235 Annex H Required ...................................................708 H.245 DTMF Signal Tone Duration (msec) ........................708 H.248 Gateways ................................................................ 449 H.320 Conversion
Attendant Console ................................................ 66, 771 H.320 Desktop Video Conferencing .............................71, 776 H.323 IP Endpoints ............................................................464 H.323 Link Bounce Recovery ............................................ 464 H.323 Security Profiles ...................................................... 468 H.323 Station Outgoing Direct Media ................................ 708 H0 .............................................................................. 545, 665 H11 .............................................................................545, 665 H12 ............................................................................ 545, 665 H1 Handover ......................................................................865 H2 Handover ......................................................................865 Handset Expander Enabled ...............................................876 Headset ............................................................................. 782 Hear Zip Tone Following VOA? ......................................... 325 Held Call UCID .................................................................. 541 Hide Post-Dialing DTMF On List Trace ..............................693 High-available cluster type .................................................586 Hold Time (min) ................................................................. 595 Holiday After Coverage ......................................................174 Holiday Coverage .............................................................. 174 Holidays ....................................................................... 94, 846 Holiday Table ..................................................................... 174 Home ................................................................................. 779 Hop Lmt ............................................................................. 686 Hospitality ............................................................ 65, 377, 834 Hospitality (G3V3 Enhancements) .....................................835 Host ........................................................................... 493, 495 Hot Desking Enhancement Station Lock ........................... 293 HOT LINE DESTINATION ................................................... 73 Hour ................................................................................... 202 Housekeeper Information Configuration ............................ 378 Housekeeping Status (Client Room) Access Code ........... 277 Housekeeping Status (Station) Access Code ....................277 Howler After Busy .............................................................. 822 Hundreds Select .................................................................. 68 Hundreds Select Button Assignments
Attendant Console ........................................................ 68 hunt group ..........................................................................388 Hunt Group:Service Level Target (% in sec) ......................405 Hunt Group Busy Activation/Deactivation ..........................262
Hunt-to Station ...................................................................745 hyperactive registration alarms ..........................................481 Hyperactive Registration Window (minutes) ......................481
I
IAA ..................................................................................... 287 IAS (Branch) ...................................................................... 162 IAS Att. Access Code ........................................................ 162 IAS Tie Trunk Group No. ....................................................162 IC Adjunct Routing .............................................................187 ICHT .................................................................................. 420 Identity for Calling Party Display ........................................979 Identity When Bridging
Feature-related system parameters ........................... 322 Idle ..................................................................................... 358 Idle/Active Ringing (Callmaster) ........................................ 771 Idle Appearance Preference .............................................. 788 Idle Code ................................................................... 228, 709 Idle Traffic Interval (seconds) .............................................464 ID Range Start/End ..........................................................1009 IGAR .......................................................................... 301, 471 IGAR Over IP Trunks ......................................................... 302 Ignore Connectivity in Server Arbitration ........................... 492 Ignore Network Answer Supervision ..................................815 Ignore Rotary Digits ........................................................... 771 Immediate .................................................................. 170, 624 Immediate Access Code ....................................................281 Immediate Redirection on Receipt of PROGRESS Inband Information ......................................................................... 811 IMS Enabled ...................................................................... 709 INADS ................................................................................695 in-band tone .......................................................................979 Inc Attd Call Record ...........................................................107 Incoming ............................................................................ 608 Incoming/Outgoing .............................................................612 Incoming Backward Signal Types (Tones to CO) .............. 615 Incoming Backward Signal Types (Tones to CO), Group A ............................................................................................ 615 Incoming Backward Signal Types (Tones to CO), Group B ............................................................................................ 616 incoming budget ................................................................ 909 Incoming call budget high threshold .................................. 910 Incoming call budget low threshold ....................................910 Incoming Call Handling Treatment .....................................420 Incoming Calling Number - Delete .....................................928 Incoming Calling Number - Format ............................510, 928 Incoming Calling Number Insert .........................................511 Incoming Calling Number - Insert ...................................... 929 Incoming Call Type ............................................................ 603 Incoming Destination ......................................................... 910 Incoming Dial Guard (msec) .............................................. 957 Incoming Dialog Loopbacks ...............................................709 Incoming Dial Tone ............................................................ 929 Incoming Dial Type ............................................................ 929 incoming digital PPM per country protocol ........................ 235 Incoming Disconnect (msec) ............................................. 957
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Avaya Aura® Communication Manager Screen Reference
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Index
Incoming Disconnect Send (msec) .................................... 958 Incoming Forward Signal Types (Tones from CO) .............613 Incoming Forward Signal Types (Tones from CO), Group I ............................................................................................ 613 Incoming Forward Signal Types (Tones from CO), Group II ............................................................................................ 614 Incoming Glare Guard (msec) ........................................... 958 Incoming Incomplete Dial Alarm (sec) ............................... 958 Incoming LDN Extension ................................................... 467 Incoming Number Format .................................................. 100 Incoming Partial Dial (sec) .................................................958 Incoming Rotary Timeout (sec) ..........................................930 Incoming Seizure (msec) ................................................... 959 Incoming Tone (DTMF) ANI ............................................... 942 Incomplete Dial Timer (sec) ............................................... 603 Increment (Start) ................................................................206 Index .................................................................................. 654 Individual Attendant Access ...............................................168 Inflow Threshold (sec) ....................................................... 401 Information Transfer Capability ..........................................687 Initial IP-IP Direct Media .................................................... 710 Insert ....................................................................................99
Call Type Analysis Table ...............................................90 Incoming Call Handling Treatment ............................. 421 Insert Digits Uniform Dial Plan Table .............................................. 984 Inserted Digits ............................................................588, 686 InSite Knowledge Base ....................................................1018 Instance ............................................................................. 861 integrated announcement .................................................... 60 Integrated Announcement Boards ..................................... 424 Integrated Announcement Extension .................................384 Integrated Announcement Translations ............................. 426 INTEGRATED CSU OPTIONS .......................................... 242 Intercept Treatment on Failed Trunk Transfers ..........325, 356 Intercom Group ..................................................................428 Interconnect ....................................................................... 228 Interdigit Pause ..................................................................822 Inter-Exchange Carrier (IXC) Codes ..................................430 Interface .............................................................228, 553, 710 ISDN BRI Trunk Circuit Pack ......................................549 Interface Channel .............................................................. 667 Interface Companding ........................................229, 549, 711 Interface Link ..................................................................... 668 Interflow-qpos EWT Threshold .......................................... 332 Interflow VDN Best Service Routing .................................................... 83 Inter-Gateway Alternate Routing ............................... 301, 467 Inter-Location .....................................................................223 Inter-location Loss Group .................................................. 575 Internal Alert ...................................................................... 942 Internal Auto-Answer of Attd-Extended/Transferred Calls . 287 Internal Coverage .............................................................. 591 International Access Code .........................................323, 571 International Call Routing .................................................. 322 International CPN Prefix .................................................... 317 Inter-PBX Attendant Service (IAS) .....................................158
Interposition ....................................................................... 168 Inter-region IP-IP Direct Audio ...........................................461 Interruptible ........................................................................672 Interruptible Aux .................................................................409 Interruptible Aux Deactivation Threshold
Hunt group ..................................................................409 Interruptible Aux Notification Timer (sec) ...........................342 Interruptible Aux threshold .................................................409 Interruptible Aux Threshold
Hunt group ..................................................................409 Inter-System IP DTMF Transmission Mode .......................481 INTERVAL 1, 2, 3 ...............................................................890 Interval between OPTIONS messages (seconds) ............. 725 Interval For Applying Periodic Alerting Tone ......................315 Intervening-regions ............................................................ 471 Interworking Feat-flag ........................................................ 107 Interworking Message ........................................ 229, 554, 711 Interworking of ISDN Clearing with In-Band Tones
Trunk Group Protocol Variations .................................979 Interworking with DCS ....................................................... 849 Intra-Location .....................................................................223 Intra-region IP-IP Direct Audio ...........................................461 Intra-Switch CDR ....................................................... 107, 432 Intra-System IP DTMF Transmission Mode .......................482 Intrusion Tone .................................................................... 307 Invalid Number Dialed Display ...........................................356 Invalid Number Dialed Intercept Treatment ....................... 357 In-VDN Time ...................................................................... 541 Invoke ID for USNI Calling Name
Trunk Group ................................................................942 inward restriction override ..................................................128 IP Address ................................................... 80, 475, 493, 495
Survivable Processor ..................................................799 IP Address Mapping .......................................................... 433 IP Attendant Consoles ....................................................... 835 IP Audio Hairpinning .................................... 69, 355, 361, 462
Signaling Group .......................................................... 711 Station ........................................................................ 771 IP codec Set ...................................................................... 435 IP Control ...........................................................................492 IP DTMF transmission mode ............................................. 482 IP DTMF Transmission Mode ............................................ 481 IPEI .................................................................................... 780 IP Interfaces .......................................................................449 IP Media Parameters ......................................................... 435 IP network region .......................................................459, 465 IP Network Region .............................................................459 IP network region page 3 ...................................................467 IP Node Names ................................................................. 475 IP parameter emergency location ...................................... 117 IP Parameter Emergency Location .................................... 118 IP Phone Group ID ............................................................ 779 IP Routing .......................................................................... 488 IP Server Interface (IPSI) Administration page 1 ...............491 IP Server Interface Administration ..................................... 491 IP Services .........................................................................498 IPSI administration .............................................................491
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Index
IPSI Connection Up Time (min) ......................................... 863 IP Softphone ...................................................................... 745 IP Softphones
video capable ............................................................. 826 IP Stations ......................................................................... 835 IP Trunks ............................................................................835 IPv4 Server Port ................................................................ 466 IPV6 Server Port ................................................................466 IP Video ..................................................................... 712, 746 IP Video Softphone ............................................................ 745 IQ (appl ccr) ....................................................................... 339 ISDN Access Code ............................................................ 262 ISDN-BRI Trunk Circuit Pack .............................................548
TN2185 circuit pack ....................................................548 ISDN-BRI Trunk Circuit Pack page 2 .................................553 ISDN-BRI Trunk Pack
TN556B, TN2198 circuit packs ...................................552 ISDN-BRI Trunks ............................................................... 835 ISDN Caller Disp ........................................................395, 882 ISDN Feature Plus .............................................................836 ISDN Network Call Redirection ..........................................836 ISDN Network Facilties ......................................................630 ISDN Parameters ...............................................................316 ISDN Precedence Call Timeout (sec) ................................ 624 ISDN-PRI ........................................................................... 836 ISDN-PRI Layer 3 public-access connections ................... 233 ISDN Protocol ...................................................................... 20 ISDN Trunk Group ............................................................. 504 Italian DCS Protocol .......................................................... 328 Italian Protocol Enabled .....................................................329 ITC ..................................................................... 191, 687, 911
Access Endpoint ...........................................................37 ITC (Information Transfer Capability) .................................746 ITN-C7 Long Timers .......................................................... 229 IXC .....................................................................................687 IXC Access Number .......................................................... 430 IXC Code Format ...............................................................430 IXC codes .......................................................................... 430 IXC Name .......................................................................... 430 IXC Prefix ...........................................................................431
J
Jack ................................................................................... 783 Jitter (ms) ...........................................................................457 Journal/Schedule Endpoint ................................................378
K
Keep-Alive Count ...............................................................465 Keep-Alive Interval (seconds) ............................................465 Keep Bridged Information on Multiline Displays During Calls ............................................................................................ 364 Keep Held SBA at Coverage Point .................................... 812 KYBD Dialing ..................................................................... 196
L
Label Language System List ................................................................... 33
LABELS FOR 2420/4620 STATIONS System List ................................................................... 34
LAI ....................................................................................... 94 Language translations ....................................................... 556 Language Translations
Automatic Wakeup ......................................................556 button labels ............................................................... 557 Enhanced Abbreviated Dialing ................................... 557 Leave Word Calling .................................................... 558 Malicious Call Trace ................................................... 558 miscellaneous call identifiers ...................................... 558 Miscellaneous features ...............................................559 Property Management Interface ................................. 559 Self Administration ......................................................559 Softkey Labels ............................................................ 560 Transfer Conference ...................................................560 View Buttons ...............................................................561 Vustats ........................................................................561 LAR ....................................................................................687 Last Board Location Saved ................................................425 Last Number Dialed Access Code .....................................263 Layer 1 Stable ....................................................................550 Layer 1 timer ......................................................................822 Least Occupied Agent ....................................................... 844 Leave Word Calling ......................................................46, 558 Leave Word Calling Cancel A Message ............................ 264 Leave Word Calling Message Retrieval Lock .................... 263 Leave Word Calling Message Retrieval Unlock .................263 Leave Word Calling Send A Message ............................... 264 Len .............................................................................100, 984 Length ................................................................................988 CDR System Parameters ............................................116 Length of Dial Code ........................................................... 429 Length of Time to Remain Connected to Announcement .. 384 Level .................................................................................. 497 Level 1 Code ......................................................................319 Level 1 Threshold (sec) ..................................................... 402 Level 2 Code ......................................................................320 Level 2 Threshold (sec) ..................................................... 402 Level of Tone Detection ..................................................... 326 Limit ................................................................................... 850 Limit Number of Concurrent Calls Activation/Deactivation 264 Line Appearance Conferencing ......................................... 313 Line Coding ........................................................................230 Line Compensation ............................................................230 Line Intercept Tone Timer .................................................. 307 Line Length ........................................................................ 930 Line Load Control .............................................................. 131 Line Load Control Restriction Level ...................................624 Lines Per Page .................................................................. 300 Linkage .............................................................................. 175 Link Failure ................................................................ 476, 480 Link Loss Delay Timeout (minutes) ....................................476
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Index
Link Loss Delay Timer (minutes) ....................................... 477 Link Loss Delay Timer (sec) .............................................. 712 List1 ............................................................................. 72, 195 List1, List2, List3 ................................................................ 164 listed directory number (LDN) ............................................161 Listed Directory Numbers .................................................. 563 List Number ....................................................................... 220 Live Stream Source ............................................................. 59 Load-sharing cluster type .................................................. 586 Local Agent Preference ..................................................... 395 Local Call Preference
Agent Login ID ..............................................................53 Local Country Code ........................................................... 322 Local Cvg Subsequent Redirection/CFWD No Ans Interval (rings) .................................................................................809 Local E.164 Country Code .................................................571 Local Ext ............................................................................ 727 Local Facility ...................................................................... 485 Local Facility # ................................................................... 497 Local Node .................................................................499, 802 Local Node Name .............................................................. 885 Local Node Number ...........................................................216 Local Only ..........................................................................799 Local Port ...................................................................499, 802 Local Preferred .................................................................. 800 Local Survivable Processor ............................................... 836 Location ..................................... 460, 580, 680, 746, 794, 893
AAR Digit Conversion Table ........................... 21, 25, 212 ARS Digit Conversion Table ........................... 21, 25, 212 Call Type Digit Analysis Table ...................................... 90 DS1 Circuit Pack ........................................................ 231 ISDN BRI Trunk Circuit Pack ......................................550 primary IPSI board ......................................................493 secondary IPSI board ................................................. 496 System Parameters Customer Options ...................... 823 Location-Based Call Type Analysis Call Type Digit Analysis .............................................. 218 Location for Covered and Forwarded Calls ....................... 809 Location for Routing Incoming Calls .................................. 712 Location Name Best Service Routing .................................................... 84 Location Parameters ..........................................................568 Long Distance Access Code ...................................... 571 Time Format on Terminals .......................................... 572 Location Parameters:Off-PBX Feature Name Extension Set ............................................................................................ 572 Locations ........................................................................... 564 Lock Best Service Routing .................................................... 84 Call Vector .................................................................... 95 Lock Messages ..................................................................746 Loc Number ....................................................................... 566 Loc Parm ........................................................................... 565 Log CTA/PSA/TTI Transactions .........................................578 Log Data Values .................................................................577 Logged-In ACD Agents ......................................................846 Logged-In Advocate Agents .............................................. 847
Logged-In IP Softphone Agents .........................................847 Logged off/PSA/TTI
Coverage Path ............................................................177 logging all submission failures ........................................... 578 Logging Levels ...................................................................577 Login .......................................................................... 246, 247
IP Options ...................................................................479 Login Access Code ............................................................274 Login Administration .......................................................... 576 Login ID
Agent Login ID ..............................................................48 LoginID for ISDN Display .....................................................48 Log IP Registrations and Events ....................................... 578 Logout ................................................................................673 Logout Access Code ..........................................................274 Logout Reason Code Type ................................................ 345
Agent Login ID ..............................................................49 Log PMS/AD Transactions .................................................578 Long Distance Access Code ..............................................571 Long Holding Time (hours) ................................................ 943 Long Hold Recall Timer ..................................................... 307 Lookahead Interflow .......................................................... 844 Look Ahead Routing .......................................................... 237 Loss Group ........................................................................ 747 Loss of Carrier Disconnect ................................................ 595 Loss Plans ......................................................................... 574 Loudspeaker Paging ..........................................................579 Lower Bound (msec) ..........................................................573 LRQ Required ....................................................................712 LSC CCA-ID ...................................................................... 366 LWC ..................................................................................... 46 LWC Activation ...................................................................772 LWC Log External Calls ..................................................... 772 LWC Reception
Agent Login ID ......................................49, 413, 771, 882
M
MAC Address ...................................................................1012 Mach ID ............................................................. 208, 668, 727 Machine ID .........................................................................589 Maid status function ...........................................................142 Main Number ..................................................................... 181 Maintain SBA At Principal ..................................................812 Maintenance Call Type ...................................................... 604 MAINTENANCE PARAMETERS
DS1 Circuit Pack ........................................................ 238 Maintenance-related system parameters .......................... 583 Maintenance Tests .............................................526, 663, 943 Make (msec) ...................................................................... 966 Malicious Call Trace ...................................................121, 558 Malicious Call Trace Activation .......................................... 265 Malicious Call Trace parameters ....................................... 836 MALICIOUS CALL TRACE PARAMETERS ...................... 302 Manual-In Access Code .....................................................274 Manual Unlock allowed ......................................................891 Mapped String
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Index
Mapped String (continued) Alias Station ..................................................................56
Mapping Mode ........................................................... 747, 795 Map-to Station ................................................................... 748 Marginal / Unacceptable .................................................... 969 Marginal Threshold
-+Dev - 2804 Hz ......................................................... 970 -+Dev - 404 Hz Loss ...................................................969 --Dev - 2804 Hz .......................................................... 969 --Dev - 404 Hz Loss ....................................................969 Max - 1004 Hz Loss ....................................................970 Maximum C Message Noise .......................................970 Maximum C Notched Noise ........................................970 Min -1004 Hz Loss ......................................................970 Minimum ERL ............................................................. 970 Minimum SRL-HI ........................................................ 970 Minimum SRL-LO ....................................................... 971 Mark Users as Phone ........................................................ 980 Mask CPN/Name for Internal Calls ....................................136 Masking CPN/Name Override ........................................... 146 Match BCA Display to Principal Class of Service ..........................................................146 Matching Pattern AAR Digit Conversion Table ................................. 25, 660 ARS Digit Conversion Table ................................. 25, 660 Uniform Dial Plan Table .............................................. 985 Max AAR Digit Conversion Table ........... 22, 25, 658, 660, 893 ARS Digit Conversion Table ........... 22, 25, 658, 660, 893 Max# Chan ........................................................................ 544 Maximum Administered Ad-hoc Video Conferencing Ports ............................................................................................ 825 Maximum Administered IP Trunks ..................................... 826 Maximum Administered Remote Office Trunks ................. 826 Maximum Administered SIP Trunks ...................................826 Maximum Agent Occupancy AUX Reason Code .............. 346 Maximum Agent Occupancy Parameters .......................... 346 Maximum Agent Occupancy Percentage .......................... 347 Maximum Auto Reserve Agents ........................................ 402 Maximum Call Rate for Direct-IP Multimedia IP Codec Set .............................................................. 442 Maximum Call Rate for Priority Direct-IP Multimedia IP Codec Set .............................................................. 441 Maximum Concurrently Registered IP eCons ....................826 Maximum Concurrently Registered IP Stations ................. 826 Maximum Entries ............................................................... 634 Maximum G700 VAL Sources ............................................826 Maximum Length ............................................................... 317 Maximum Number of Avaya Media Server VoIP Channels ............................................................................................ 827 Maximum Number of Call Forwarding Hops ......................816 Maximum Number of Digits for Directed Group Call Pickup ............................................................................................ 364 Maximum Number of DS1 Boards with Echo Cancellation 826 Maximum Number of Expanded Meet-me Conference Ports ............................................................................................ 827 Maximum Number of External Calls Logged Per Station .. 291
Maximum Number of Messages Per Station ..................... 291 Maximum Number of Trunks to Use for IGAR ................... 467 Maximum Off-PBX Telephones ..........................................823 Maximum Off-PBX Telephones - OPS ............................... 824 Maximum Off-PBX Telephones -- PBFMC ........................824 Maximum Off-PBX Telephones - PVFMC .......................... 824 Maximum Off-PBX Telephones - SCCAN .......................... 824 Maximum Percentage of Trunks Which Can Be Removed From Service by ATMS ...................................................... 971 Maximum Ports per Expanded Meet-me Conf ...................313 Maximum Precedence Level ............................................. 131 Maximum Resend Requests ..............................................604 Maximum Size of UUI IE Contents .................................... 526 Maximum Stations ............................................................. 824 Maximum Suppression Time
Best Service Routing .................................................... 84 Maximum Time Agent in ACW before Logout (sec.) ..........348 Maximum time agent in ACW before logout (sec)
Agent Login ID ..............................................................49 Maximum TN2501 VAL Boards ..........................................827 Maximum TN2602 Boards with 320 VoIP Channels ..........827 Maximum TN2602 Boards with 80 VoIP Channels ............827 Maximum Video Capable IP Softphones ........................... 826 Maximum Video Capable Stations .....................................827 Maximum XMOBILE Stations ............................................ 825 Max Message Size to Send ............................................... 520 Max NCA TSC ................................................................... 555 Max number of CA TSC .....................................................713 Max number of NCA TSC .................................................. 713 Max Ports .........................................................................1009 may-have-extra-digits ........................................................ 981 MCSNIC .............................................................................146 mct-control ......................................................................... 249 MCT group extensions, 1 to 100 ........................................249 MCT Voice Recorder Trunk Group .................................... 302 Measured ...................................................403, 527, 943, 992 Media Complex Ext ............................................................748 Media Encryption ....................................................... 437, 713 Media Encryption Over IP ..................................................837 Media-Gateway ..................................................................584 Media Parameters ............................................................. 461 Media Server ..................................................................... 585 media server recovery rule
Communication Manager ........................................... 856 Meet-me Conf ...................................................................... 95 Meet-me Conference ......................................................... 993 meet-me Conference Access Code Change ..................... 265 Member Assignment Method ............................................. 511 Member Range Allowed .................................................... 418 Message Center ................................................................ 413 Message Center AUDIX Name ..........................................413 Message Center ID ............................................................589 Message Center MSA Name ............................................. 414 message lamp ....................................................................117 Message Lamp ...................................................................118 Message Lamp Ext ............................................................ 748 Message Sequence Trace (MST) Disable ......................... 265
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Index
Message Server Name ...................................................... 789 Message Waiting Configuration ......................................... 379 Message Waiting Indication for External Calls ...................291 Message Waiting Indication Subscriber Number Prefixes . 588 Message Waiting Indicator .................................................748 Message Waiting Lamp Indicates Status For .................... 330 Message Waiting Type .......................................................749 Messaging Server Name for Messaging
Agent Login ID ..............................................................50 Metric ................................................................................. 490 MF ANI Prefix .................................................................... 127 MFE Type .......................................................................... 605 MF Incoming Call Trace .....................................................131 MF Interdigit Timer (sec) ....................................................573 MF Signaling Intercept Treatment - Incoming ....................604 MF Signaling Intercept Treatment - Outgoing ....................605 MF Tariff Free .....................................................................944 MGR1 ................................................................................ 695 MIA Across Skills
Agent Login ID ..............................................................50 MIA Splits or Skills ............................................................. 337 Migrate H.248 MG to primary ............................................ 854 Milliseconds Before PMS Link Acknowledgment Timeout . 380 MIM Mtce/Mgt ............................................................ 200, 749 MIM Support ...................................................................... 200 MIM Support (Management Information Message Support) ............................................................................................ 749 Min
AAR Digit Conversion Table ........... 22, 26, 658, 661, 893 ARS Digit Conversion Table ........... 22, 26, 658, 661, 893 Min# Chan ......................................................................... 544 Minimum Agent-LoginID Password Length ................. 51, 330 Minimum Digit Length ........................................................ 323 Minimum Station Security Code Length ............................ 697 Minimum time of network stability ......................................855 Minute ................................................................................ 202 Miscellaneous Call .............................................................168 Miscellaneous features ...................................................... 559 Miscellaneous Parameters .................................................. 43 Misoperation Alerting ......................................................... 326 MLPP Service Domain .......................................................132 MM (WSM) Route Pattern ................................................. 865 MM Early Answer ...............................................................395 MMI Cabling Board ............................................................ 231 MMI Interface .....................................................................231 Mobile Call (CTI) Extension ............................................... 639 Mobility Trunk Group ..........................................................749 Mode ..........................................................................668, 974 Mode Code for Centralized Voice Mail .............................. 837 Mode Code Interface ......................................................... 308 Mode code related system parameters ............................. 590 Mode Codes (From Switch to VMS) .................................. 590 Model ................................................................................. 750 Modem Mode IP codec set ................................................................442 Modem Name .................................................................... 595 Modem Pool Group ............................................................593
modem pool group number ................................................594 modem pool group type ..................................................... 594 Modem Relay Preferred (NoAudio) ................................... 445 Modified Circuit ID Display .................................................108 Modify Tandem Calling Number .........................................527
Trunk Group ................................................................944 Modulation Mode ............................................................... 445 MOH Group ....................................................................... 599 MOH Group Name .............................................................600 MOH Source Location ....................................................... 599 Month .................................................................................202 Month (Start) ......................................................................206 Month (Stop) ...................................................................... 206 More Members Exist ..........................................................418 More VDN's ........................................................................245 Mounting ............................................................................ 783 MSA Names .........................................................................80 Mtce ................................................................................... 472 Multifrequency Signaling ....................................................837 Multifrequency-Signaling-Related Parameters ....................................................................601, 609, 612, 617 Multifrequency-Signaling-Related Parameters page 1 ...... 601 Multimedia ........................................................................... 95 Multimedia Appl. Server Interface (MASI) ......................... 837 Multimedia Call Access Code ............................................ 279 Multimedia Call Handling (Basic) .......................................837 Multimedia Call Handling Enhanced ..................................837 Multimedia Data Conference Activation .............................279 Multimedia Data Conference Deactivation ........................ 279 Multimedia Early Answer ................................................... 772 Multimedia IP SIP Trunking ............................................... 837 Multimedia Multi-Address Access Code ............................ 279 Multimedia Parameter Access Code ................................. 280 Multinational Locations ...................................................... 838 Multiple Call Handling ................................................ 403, 844 Multiple Call Handling (Forced) ......................................... 844 Multiple Level Precedence .................................................132 Multiple Level Precedence & Preemption (MLPP) Parameters ............................................................................................ 621 Multiple Level Precedence and Preemption ...................... 838 Multiple Level Precedence and Preemption (MLPP) ......... 280 Multiple Locations .............................................................. 838 MUSIC/ANNOUNCEMENTS IP-CODEC PREFERENCES ............................................................................................ 485 Music/Tone on Hold ........................................................... 287 Music (or Silence) On Transferred Trunk Calls ..................287 Music on Hold .................................................................... 127 Music Source ..................................................................... 871 Music Sources ................................................................... 627 Mute Button Enabled ......................................................... 749 MWI - Number of Digits per Voice Mail ..............................317 MWI Served User Type ................................................50, 772 My Docs ...........................................................................1016
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Index
N
Name .... 546, 563, 566, 630, 651, 652, 654, 663, 974, 993, 1009
Administered Connection ............................................. 41 Agent Login ID ..............................................................50 Attendant Console ............................................ 67, 72, 74 Call Vector .................................................................... 96 Coverage Answer Group ............................................ 172 CTI Link ...................................................................... 185 Data Module ............................................................... 191 DS1 Circuit Pack ........................................................ 232 Duplicate Vector ......................................................... 245 Group Paging Using Speakerphone ...........................372 Holiday Table .............................................................. 375 Hunt Group ................................................................. 419 Intercom Group ...........................................................429 IP Network Region ......................................................460 IP Node Names .......................................................... 475 ISDN BRI Trunk Circuit Pack ......................................550 Network Region Group ............................................... 631 Signaling Group ..........................................................714 Station ........................................................................ 750 Terminating Extension Group ..................................... 883 National CPN Prefix ...........................................................317 NCA-TSC Trunk Member ...................................................527 nc Trk Call Splitting ............................................................ 107 Near-end CSU ................................................................... 239 Near End Establishes TCP Signaling Socket .................... 469 Near-end Listen Port ..........................................................714 Near-end Node Name ........................................................714 Near End TCP Port Max .................................................... 469 Near End TCP Port Min ..................................................... 469 Net ............................................................................. 661, 985 AAR Digit Conversion Table ......................................... 26 ARS Digit Conversion Table ......................................... 26 NET ....................................................................................695 Net Redir Best Service Routing .................................................... 84 Network (Japan) Needs Connect Before Disconnect ........ 528 Network Bits .......................................................................490 Network Call Redirection ................................... 528, 689, 980 Network Call Transfer ........................................................ 715 Network Facilities ...............................................................630 Network Feedback During Tone Detection ........................ 326 Network Level .................................................................... 320 Network Management Protocol ......................................... 241 Network region ...................................................................451 Network Region ......................................................... 434, 680 Network Region Group ...................................................... 631 Network uses 1's for Broadnet Addresses .........................452 Next ANI Digit .................................................................... 608 Next Path Number ............................................................. 175 Night .......................................................................... 546, 974 night bells ...........................................................................271 Night Destination ....................................................... 563, 871
Night Serv .......................................................................... 422 Night Service ................................................................64, 911 Night Service Act. Ext. ....................................................... 162 Night Service Destination .................................................. 396 Night Service Disconnect Timer .........................................308 nite-serv ............................................................................... 64 No. Del. Digits ....................................................................688 No. Dgts Subaddress .........................................................688 No. of Calls ........................................................................ 676 No Answer Timeout (sec) .................................................. 165 Node Name ........................................................452, 456, 680
Survivable Processor ..................................................800 Node Num ..........................................................................985 Node Number .................................................................... 632
AAR Digit Analysis Table .............................................. 22 ARS Digit Analysis Table .............................................. 22 Node Number (Local PBX ID) ............................................108 Node Number Routing ....................................................... 632 No Dial Tone Conferencing ........................................312, 313 No Hold Conference Timeout ............................................ 313 non-SCCAN applications ................................................... 931 Normal Outgoing Seize Send (msec) ................................ 959 No Service Time Out Interval .............................................863 notify Hold notification .......................................................... 941 Unhold notification ...................................................... 941 NPA ....................................................................................689 NR Group IP Network Region ......................................................459 Num Best Service Routing .................................................... 85 Number ...................................................................... 655, 994 Best Service Routing .................................................... 85 Call Vector .................................................................... 96 Holiday Table .............................................................. 375 Service Hours Table ................................................... 699 Numbering Format .............................................................944 Route Pattern ............................................................. 689 Numbering -- Public/Unknown Format ............................. 636 Numbering-Public/Unknown format ..................................... 69 Number of Codes Administered ...........................................82 Number of Digits from PMS ............................................... 385 Number of Digits in PMS Coverage Path .......................... 385 Number of Emergency Calls Allowed in Attendant Queue 297 Number of Housekeeper ID Digits ..................................... 379 Number of Incoming ANI Digits ......................................... 610 Number of Members .......................................................... 511 Trunk Group ................................................................912 Number of OPTIONS message failure before failover .......725 Number of OPTIONS message successes before failback ............................................................................................ 725 Number of Outgoing ANI Digits ......................................... 610 Number of Pings Per Measurement Interval ......................478 Number of Recordings .......................................................425 Number of Registrations within the Window ...................... 481 Number of Rings ................................................................177 NxDS0 ............................................................................... 545
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Index
NXDS0 ...............................................................................665
O
Object ................................................................................ 497 Observe on Agent Answer .................................................999 Off ...................................................................................... 592 Off-Hook Alert .................................................................... 143 Offline Call Logging ............................................................780
Station ........................................................................ 780 Off-Net Cvg Subsequent Redirection/CFWD No Ans Interval (rings) .................................................................................810 Off-PBX Feature Name Extension Set ...............................572 Off-PBX telephone configuration set ..................................151 Off-PBX telephone feature name extensions .....................250 Off-PBX Telephone Mobile Feature Extensions .................639 Off-PBX telephone station-mapping .................................. 790 Off Premises Station .......................................................... 751 Off-Premises Tone Detect Timeout Interval ....................... 288 One-Step Recording .......................................................... 314 one-X Server Access ......................................................... 136 On-hook Dialing on 607/2400/4600/6400/8400 Terminals .326 Options .............................................................................. 349 OPTIONS Request Parameters .........................................725 Originating Auto Restoration ..............................................663 Originating Extension .................................................181, 693 Originator
Administered Connection ............................................. 41 Other LAI Information ........................................................ 541 Other Related Parameters .................................................591 Other Stations When Call Is Active ....................................358 Other Stations When Call Is Put On-Hold ..........................359 Outg Attd Call Record ........................................................108 Outgoing ............................................................................ 609 Outgoing ANI ..................................................................... 945 Outgoing Backward Signal Types (Tones from CO) .......... 618 Outgoing Backward Signal Types (Tones from CO), Group A ............................................................................................ 618 Outgoing Backward Signal Types (Tones from CO), Group B ............................................................................................ 619 Outgoing Budget ................................................................912 Outgoing call budget high threshold .................................. 913 Outgoing call budget low threshold ....................................913 Outgoing Call Type ............................................................ 605 Outgoing Channel ID Encoding ......................................... 528 Outgoing Dial Guard (msec) .............................................. 960 Outgoing Dial Type .................................................... 647, 930 Outgoing Disconnect (msec) ............................................. 960 Outgoing Disconnect Send (msec) .................................... 960 Outgoing Display ............................................... 512, 643, 914 Outgoing End of Dial (sec) .................................................960 Outgoing Forward Signal Absent Timer (sec) ....................605 Outgoing Forward Signal Present Timer (sec) ...................606 Outgoing Forward Signal Types (Tones to CO) ................. 617 Outgoing Forward Signal Types (Tones to CO), Group I ... 617 Outgoing Forward Signal Types (Tones to CO), Group II .. 618 Outgoing Glare Guard (msec) ........................................... 961
Outgoing II by COR ........................................................... 610 Outgoing Last Digit (sec) ................................................... 961 Outgoing Number Format .................................................. 101 Outgoing Rotary Dial Interdigit (msec) ...............................962 Outgoing Seizure (msec) ................................................... 962 Outgoing Seizure Response (sec) .....................................963 Outgoing Shuttle Exchange Cycle Timer (sec) .................. 573 Outgoing Start Timer (sec) .................................................606 Outgoing Trunk Alerting Timer ........................................... 132 Outgoing Trunk Disconnect Timer ............................. 132, 136 Outgoing Trunk Groups
Calling Party Number Conversion for Tandem Calls .. 101 Outg Trk Call Splitting ........................................................ 108 Outpulse Without Tone ...................................................... 326 Outpulsing Information .......................................................965 Overlap Sending on Link-to-Link Tandem Calls .................606 Overview timer to Group Queue ........................................ 166
P
PA Agent Login ID ..............................................................53
Packet loss (%) ..................................................................457 Packet Loss (%) .................................................................478 Packet Resp Timer .................................................... 501, 804 Packet Size ........................................................................443 Packet Size (ms) ................................................................439 Pager Number ................................................................... 181 Parameters for creating QSIG selection numbers ............. 319 Parameters for Media Gateway Alarms .............................481 Parameters for Network Region Registration (NR-REG) Alarms ................................................................................481 Parity ..................................................................................198 Partitioned Group Number .........................................127, 632 Partition Routing Table .......................................................640 Passive Signalling Station ................................................. 752 Passphrase ........................................................................715 Pass Prefixed CPN to ASAI ...............................................318 Pass Prefixed CPN to VDN/Vector .................................. 1000 Password ........................................................... 246, 247, 502
Agent Login ID ..............................................................51 IP Options ...........................................................479, 483 Password enter again Agent Login ID ..............................................................51 Password to Change COR by FAC ....................................354 PASTE (Display PBX Data) ............................................... 844 PASTE Access Code ......................................................... 265 PASTE Display PBX Data on telephone Class Of Restriction ....................................................127 Path Replacement ............................................................. 536 Path Replacement Method ................................ 529, 536, 945 Path Replacement While in Queue/Vectoring ....................318 Path Replacement with Measurements ............................. 318 Path Replacement with Retention ............................. 537, 945 Pattern Choices ................................................................. 682 Pattern Name .....................................................................689 Pattern Number ................................................................. 690
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Index
Pause (msec) .....................................................181, 539, 965 Pause Duration .................................................................. 729 PBX ID ............................................................................... 946 PCN notification ............................................................... 1020 PC Non-Predictive Reports Skill ........................................ 342 PCOL/TEG Call Alerting .................................................... 773 PC Predictive Reports Skill ..............................................1000 Peer Detection Enabled .....................................................715 Peer Protocol ..................................................................... 232 Peer Server ........................................................................715 Per Button Ring Control .....................................................773 Per Call CPN/BN ............................................................... 421 Per Call CPN Blocking Code ............................................. 946 Per Call CPN Blocking Code Access Code
Feature access codes ................................................ 265 Per Call CPN Unblocking Code ......................................... 946 Per Call CPN Unblocking Code Access Code
Feature access codes ................................................ 266 Percentage IPBO Blocked ................................................. 658 Percent Full ................................................................212, 985
AAR Digit Conversion Table ................... 22, 26, 658, 893 ARS Digit Conversion Table ................... 22, 26, 658, 893 Period ................................................................................ 655 Periodic Registration Timer (min) ...................................... 477 Permanently Disable ..........................................................676 Permit Mismatch ................................................................ 198 Personal CO Line Group ........................................... 641, 651 Personalized Ringing Pattern ............................................ 751 Personal List Abbreviated Dialing ...................................................... 31 List Number .................................................................. 31 Personal List .................................................................32 Personal Station Access Processor Ethernet .....................................................839 Personal Station Access (PSA) ................................. 144, 838 Personal Station Access (PSA) Associate Code ............... 266 Personal Station Access (PSA) Dissociate Code .............. 266 Per Station CPN - Send Calling Number ........................... 773 PGN/TN/COR for Covered and Forwarded Calls .............. 810 PGN # ................................................................................ 888 PGN 1 (through PGN 8) .....................................................640 Phantom Calls ................................................................... 850 phone number ....................................................................790 Phone Number Stations With Off-PBX Telephone Integration .............792 Pickup Group ..................................................................... 652 Pickup Group Number ....................................................... 248 Pickup Number .................................................................. 248 Pickup on Transfer .............................................................359 PIN Checking for Private Calls .......................................... 364 PIN Checking for Private Calls Access Code .................... 268 PIN Checking for Private Calls Using AAR Access Code Feature Access Codes ............................................... 269 PIN Checking for Private Calls Using ARS Access Code Feature Access Codes ............................................... 269 pings per measurement interval ........................................ 478 Ping Test Interval (sec) ...................................................... 478
Pin Number ........................................................................182 Plan # .................................................................................543 Platform
System Parameters Customer Options ...................... 825 Platform Maximum Ports ................................................... 825 PMS Endpoint ....................................................................380 PMS Link Maximum Retransmission Requests .................380 PMS Link Maximum Retransmissions ............................... 380 PMS Log Endpoint .............................................................379 PMS Protocol Mode ...........................................................380 PMS Sends Prefix ..............................................................385 PN
system parameters port networks .............................. 862 PNC duplication ................................................................. 852 PNC Duplication ................................................................ 838 PN Cold Reset Delay Timer (sec) ......................................864 Point1, Point2, ... ...............................................................177 Policy Routing Table .......................................................... 654 polling feature .................................................................... 349 Port .................................................... 547, 550, 555, 580, 975
Attendant Console ........................................................ 67 Data Module ............................................................... 192 Feature-Related System Parameters ......................... 288 Station ........................................................................ 751 Port # ................................................................................. 497 IP Options ...................................................................485 Port Board Security ............................................................696 Port Board Security Notification Interval ............................ 696 Port Extension Agent Login ID ..............................................................51 port matrix ........................................................................1016 Port Network Support ........................................................ 839 Port Pair Assignments ....................................................... 598 Post Connect Dialing Options ............................................157 Posted Messages ...................................................... 266, 839 PPM ................................................................................... 966 PPM per country protocol .................................................. 235 PPS ....................................................................................967 Precedence Calling Access Code ..................................... 280 Precedence Calling-Dialed Digit Assignment .................... 623 Precedence Call Timeout (sec) ..........................................625 Precedence Call Waiting ....................................................774 Precedence Incoming ........................................................ 946 Precedence Outgoing ........................................................ 946 Precedence Routing Digit Analysis Table .......................... 657 Precedence Routing Digit Conversion Table ..................... 660 Preemptable ...................................................................... 132 Preempt Emergency Call ...................................................625 Preempt Method ................................................................ 658 Prefer H.323 over SIP for Dual-Reg station 3PCC Make Call ............................................................................................ 342 Preferred Minimum Session Refresh Interval sec Trunk group ................................................................ 931 Prefer use of G.711 by Announcement Sources ................485 Prefer use of G.711 by IP Endpoints Listening to Announcements .................................................................485 Prefer use of G.711 by IP Endpoints Listening to Music ....485
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Index
Prefer use of G.711 by Music Sources ...................... 480, 486 Prefix ..........................................................................566, 984 Prefix-1 ...................................................................... 648, 914 Prefix Mark .........................................................................690 Prepend '+' to Calling/Alerting/Diverting/Connected Number
Trunk Group ................................................................980 Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers ................................................................. 716 Prgm Group List Access Code .......................................... 254 PRI Endpoint ......................................................................662 Primary
Hunt Group ................................................................. 414 Primary D Channel ............................................................ 716 Primary IPSI .......................................................................492 PRIMARY LEVELS ............................................................ 878 Primary MFSS CCA-ID ...................................................... 366 primary output endpoint ..................................................... 108 Primary Output Format ...................................................... 109 Primary Search Time (seconds) ........................................ 477 Priority ........................................................................170, 625
Administered Connection ............................................. 44 Priority Access Code ......................................................... 281 Priority Calling ....................................................................144 Priority Calling Access Code ............................................. 267 Priority Factory Number ...................................................1009 Priority Ip Video ..................................................................147 Priority On Intraflow ........................................................... 404 Priority Queuing ................................................................. 128 Priority Ring ....................................................................... 136 Priority Score ..................................................................... 800 Priority Video ......................................................................716 Priority with respect to Media Servers ............................... 802 Privacy -- Attendant Lockout ............................................ 159 Privacy -- Digits to Hide .................................................... 109 Private Group II Permissions and Public Interworking .......607 Private Networking .............................................................839 Private Prefix ..................................................................... 634 Privileged
Enhanced List ...................................................28, 30, 34 Personal List .....................................................28, 30, 34 System List ....................................................... 28, 30, 34 Proc Chan ..........................................................................668 Processor and System MSP ..............................................839 Processor Channel Assignment/ Communication Interface processor ........................................................................... 666 Processor Ethernet Network ..............................................800 Product ID ..........................................................................850 Program Access Code .......................................................267 Program Ext Group List ..................................................................... 30 Programmed Dial Pause (msec) ................................539, 963 Prohibit Bridging Onto Calls with Data Privacy ..................291 Prompting ............................................................................ 96 Prompting Timeout (secs) ..................................................332 propagation delay .............................................................. 479 Property Management Interface ........................................ 559 Protected
Protected (continued) Announcements Audio Sources ................................... 60
Protocol for Caller ID Analog Terminals .............................288 Protocol Type .....................................................................914 Protocol Variations ............................................................. 976 Protocol Version .........................................................232, 717 Provide forced local ringback for EC500
Configuration Set ........................................................157 Provide Ringback ...............................................................414 Proxy Sel Rte Pat .............................................................. 567 PSA display information .....................................................293 PSN notification ............................................................... 1020 Public (PSTN) Number ...................................................... 789 public-network access connections for ISDN-PRI Layer 3 233 public network trunks conference call ................................308 Pull Transfer .......................................................................327
Q
QoS and Ethernet Settings ................................................ 494 QSIG/ETSI TSC Extension ................................................318 QSIG and SIP Diverted Calls Follow Diverted to Party's Coverage Path ...................................................................810 QSIG Call Offer Originations ..............................................144 QSIG CAS Number ............................................................162 QSIG Path Replacement Extension .................................. 318 QSIG Reference Trunk Group ........................................... 981 QSIG selection numbers, parameters for creating ............ 319 QSIG Signaling Group ....................................................... 717 QSIG to DCS TSC Gateway ..............................................670 QSIG Trunk Group Options ............................................... 535 QSIG VALU Coverage Overrides QSIG Diversion with Rerouting ........................................................................... 812 QSIG Value-Added ............................................................ 537 QSIG-Value Coverage Encoding ....................................... 537 Q-SIP ................................................................................. 717 Queue .................................................................................. 61
Hunt Group ................................................................. 396 Queue Length ............................................................ 512, 914
Announcements Audio Sources ................................... 61 Queue Limit ....................................................................... 396 queue priorities .................................................................. 167
R
R2 MFC Signaling ..............................................................947 R2-MFC signaling trunks ................................................... 126 Radio Controllers with Download Server Permission ........ 354 Rate ............................................................................. 61, 425 R Balance Network
Station ........................................................................ 752 RCL ....................................................................................894 Reachability for domain control SIP Stations
enable disable ................................................................. 349
Reason Code Names ........................................................ 672 Reason Codes ................................................................... 844
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Index
Recall from VDN ................................................................ 308 Recall Rotary Digit ..................................................... 354, 774 Recall Timing ..................................................................... 573 Receive ALBO ................................................................... 242 Receive Analog Incoming Call ID ...................................... 947 Receive Answer Supervision ..................................... 649, 932 Receive Automatic Line Build-Out ..................................... 242 Receive Buffer TCP Window Size ..................................... 452 Received Digital Metering Pulse Maximum (ms) ............... 234 Received Digital Metering Pulse Minimum (ms) ................ 235 Received Digital Metering Pulse Value .............................. 235 Received Signal Gain (dB) ................................................ 607 Receive Release Ack .........................................................932 Receiver Responds to Remote Loop .................................596 Receive Space Disconnect ................................................595 Receive Unencrypted from IP Endpoints ...........................697 Record Agent ID on Incoming ............................................ 110 Record Agent ID on Outgoing ............................................ 110 Record Call-Assoc TSC ..................................................... 110 Record Called Vector Directory Number Instead of Group or Member .............................................................................. 110 Recording Delay Timer ...................................................... 315 Record Length ....................................................................116 Record Non-Call-Assoc TSC ..............................................111 Record Outgoing Calls Only ...............................................111 Recovery Rule Number ..................................................... 855 Redirected Call .................................................................. 168 Redirected DID Call ........................................................... 169 Redirection on No Answer Aux Work Reason Code ..........345 Redirect Notification ...........................................................774 Redirect on IP/OPTIM Failure to VDN ............................... 410 Redirect on No Answer (rings) ...........................................410 Redirect on No Answer to VDN ......................................... 410 Redirect on OPTIM failure ................................................. 932 Redundancy
IP Codec Set .............................................................. 443 Referral Destination ........................................................... 694 Refresh MW Lamp .............................................................591 Refresh Terminal Parameters Access Code ......................267 Region ............................................................................... 460 registration alarms ............................................................. 481 Rel ..................................................................................... 851 Release Ack Send (msec) ................................................. 963 Reliable Protocol ........................................................501, 804 Remote Access ..................................................................674 Remote Access Dial Tone ..................................................676 Remote Access Extension .................................................677 Remote Attendant Route String .........................................625 Remote Call Coverage Table .............................................678 Remote Logout of Agent ....................................................133 Remote Logout of Agent Access Code ..............................274 Remote Loop-Around Test .................................................192 Remote Node .............................................................499, 803 Remote Office .................................................... 680, 718, 839 Remote Office Phone .........................................................753 remote office trunks ........................................................... 826 Remote Port ...............................................................499, 803
Remote Send All Calls .......................................................267 Remote Softphone Emergency Calls
Attendant Console ........................................................ 70 Remote Soft Phone Emergency Calls ............................... 774 Remote VMS Extensions- First ..........................................592 Remote VMS Extensions - Second ................................... 592 Remove '+' from Incoming Called/Calling/Alerting/Diverting/ Connected Numbers ..........................................................718 Remove '+' from SIP Numbers
CDR System Parameters ............................................111 CTI Link ...................................................................... 187 Remove # From Called Number .........................................111 Remove Agent Skill Access Code ..................................... 275 Remove Inactive BCMS/VuStats Agents ........................... 340 Repetitive Call Waiting Interval (sec) ................................. 327 Repetitive Call Waiting Tone .............................................. 327 Replace International Access Code with '+' .......................574 Location Parameters .................................................. 574 Replacement String AAR Digit Conversion Table ................................. 26, 661 ARS Digit Conversion Table ................................. 26, 661 Replace Restricted Numbers .............................................529 Replace Unavailable Numbers .................................. 529, 947 report expiration Avaya Aura MS ...........................................................857 Reporting for PC Non-Predictive Calls .............................. 342 Reporting for PC or POM Calls ........................................1000 report interval .....................................................................857 Request Call Category at Start of Call ............................... 611 Request Category ..............................................................948 Request CPN at Start of Call ............................................. 611 Request Incoming ANI (non-AAR/ARS) ............................ 607 Request URI contents ........................................................981 Require Mutual Authentication if TLS ................................ 780 Resend display UPDATE once on receipt of 481 response ............................................................................................ 977 Reserved Slots for Attendant Priority Queue .....................297 Reset Shift Timer ............................................................... 309 Restart ANI from Caller Category ...................................... 611 Restrict Call Forward Off Net ............................................. 840 Restrict Call Fwd-Off Net ................................................... 144 Restrict Call Joining ........................................................... 273 Restrict Calls ......................................................................294 Restrict Customization Of Button Types .............................. 87 Restrict Customization Of Labels For the Following Button Types ................................................................................... 87 Restricted Call List .............................................................128 Restricted Number Dialed Display .....................................357 Restricted Number Dialed Intercept Treatment ................. 357 Restrict First Consult Activation Deactivation Feature Access Codes ............................................... 273 Restriction - Controlled ...................................................... 271 Restriction Override ........................................................... 128 Restrict Last Appearance .................................................. 774 Restrict Second Agent Consult ..........................................133 Restrict Second Call Consult Class of Restriction .....................................................133
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Index
Restrict Second Consult Activation Deactivation Feature access codes ................................................ 273
Retries ............................................................................... 182 Re-try ................................................................................. 355 Retry ARS/AAR Analysis if All-Location Entry Inaccessible ............................................................................................ 216 Retry Interval
Administered Connection ............................................. 44 Retry Interval (sec) ............................................................ 182 Retry upon RSVP Failure Enabled .................................... 463 Return Call .........................................................................169 Return Call Timeout (sec) .................................................. 166 Return Destination ........................................................... 1001 Reverse Star ......................................................................332 RFC 3389 Comfort Noise .................................................. 718 Rg ...................................................................................... 783 RHNPA Table ..................................................................... 682 RHNPA TABLE .................................................................. 682 Ringer Off/Auto Answer ..................................................... 341 Ringing ...............................................................................359 Ringing Monitor
Trunk Group ................................................................963 RL
Agent Login ID ..............................................................53 RLT Trunk Group No. .........................................................163 RMS Feature Enabled ....................................................... 696 Rng .................................................................................... 178 Roll-About Video ................................................................ 238 ROOF Failure and Unreachable Aux Work Reason Code .345 Room
Station ........................................................................ 783 Room Activated Wakeup with Tones ................................. 385 Room Change/Swap status function ................................. 142 Roundtrip Propagation Delay (ms) .................................... 479 Route Number
IP Routing ...................................................................490 Route Pat ...........................................................................632 Route Pattern .............................................................252, 683
AAR Digit Analysis Table .............................................. 22 ARS Digit Analysis Table .............................................. 22 Route Selection ................................................................. 786 Route-to Number ............................................................. 1004 Route-to VDN .................................................................... 655 Routine .............................................................................. 625 Routine Access Code ........................................................ 281 Routing Digits .................................................................... 589 Routing Digits (e.g. AAR/ARS Access Code) .................... 414 Routing Extension on Unavailable Voice Synthesis .......... 385 RRQ Required ................................................................... 718 RSVP Enabled ...................................................................464 RSVP Profile ......................................................................464 RSVP Refresh Rate (secs) ................................................ 464 RSVP unreserved (BBE) PHB Value ................................. 464 RTCP Monitor Server .........................................................479 RTCP Monitor Server Parameters; .................................... 466 RTCP Reporting to Monitor Server Enabled ......................465 RTCP Report Period (secs) ............................................... 466
RT Delay (ms) ....................................................................457 Rule ................................................................................... 206 Rule Name .........................................................................855 Russian signaling trunks ....................................................126
S
S8300E .............................................................. 454, 839, 981 SAC/CF Override .......................................................188, 781 SAC/CF Override by Priority Call and Dialing ....................136 SAC/CF Override by Team Btn
Class Of Restriction ....................................................137 SAC/CF Override Protection for Priority Call and Dialing .. 137 SAC/CF Override Protection for Team Btn ........................ 137 SAC Notification .................................................................164 SBS ....................................................................530, 538, 719 SCCAN ...................................................................... 691, 932 SCCAN Parameters ...........................................................323 Scheduled ..........................................................................543 Schedule Download ...........................................................368 Scope .................................................................................988 Scroll Status messages Timer (sec.) ................................. 352 SDP Capability Negotiation for SRTP ................................362 searching for content ....................................................... 1016 Second ...............................................................................203 Second Announcement Delay (sec) .................................. 415 Second Announcement Extension .....................................415 Second Announcement Recurring .....................................415 Secondary Alert on Held Reminder Calls? ........................ 166 Secondary data module .....................................................193 Secondary Data Module .................................................... 840 Secondary IPSI ..................................................................495 Secondary Output Endpoint ............................................... 112 Secondary Output Format ..................................................112 Seconds Before PMS Link Idle Timeout ............................ 380 Secure
Enable File Transfer ................................................... 246 Enable Session ...........................................................247 Secure SIP .........................................................................691 Secure Terminal Equip .......................................................756 Security Code ............................................................ 643, 753 Agent Login ID ..............................................51, 396, 882 Attendant Console ........................................................ 68 Security Code for Terminal Self Administration Required ..697 Security violation notification ..................................... 693, 696 Security violation notification parameters .................. 693, 695 Seize Ack Delay (msec) .....................................................964 Seize Ack Send (msec) ..................................................... 964 Seize When Maintenance Busy .........................................948 Selected Survivable Processor Node Names ....................723 Select Last Used Appearance ........................................... 777 Self Administration .............................................................559 Self Station Display Activation ........................................... 267 Self-Station Display Enabled ............................................. 289 Send All Call Options .........................................................302 Send All Calls Activation/Deactivation ............................... 267 Send All Calls and Call Forwarding (SAC/CF) Override ....134
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Index
Send All Calls Applies to ....................................................303 Send ANI for MFE ..............................................................128 Send ANSI-T1.403 One-Second Performance Reports .... 241 Send Answer Supervision ..................................................933 Send Called/Busy/Connected Number ...................... 530, 949 Send Calling Number .................................................530, 949 Send Codeset 6/7 LAI IE ................................................... 531 Send Connected Number .......................................... 531, 950 Send Connect Event to ASAI For Announcement Answer 343 Send Custom Messages Through QSIG? ......................... 318 Send Disconnect Event for Bridged Appearance .............. 187 Send Diversion Header ......................................................981 Send DTMF Over Telecommuter Link
Feature-related system parameters ........................... 327 Send EMU Visitor CPN ...................................................... 950 Send Incoming/Outgoing Disconnect Timers to TN465 Ports ............................................................................................ 964 Sending Delay ................................................................... 592 Send ISDN Trunk Group Name on Tandem Calls ............. 318 Send Name ................................................................532, 951 Send Non-ISDN Trunk Group Name as Connected Name 319 Send QSIG Message Center ID ........................................ 589 Send Release Ack ............................................................. 933 Send Reroute Request ...................................................... 416 Send Space Disconnect .................................................... 596 Send Transferring Party Information ..................................982 Send UCID .........................................................................533 Send UCID to ASAI ........................................................... 338 Send UUI IE .......................................................................533 Serial Call .......................................................................... 169 Server ID ....................................................................502, 801 server IP address ...............................................................480 Server IPv4 Address ..........................................................466 Server IPv6 Address ..........................................................466 server port ..........................................................................480 Service/Feature ......................................... 422, 544, 691, 727 Service Hours Table ...........................................................698 Service Interruption ............................................................622 Service Level Algorithm for SLM ....................................... 343 Service Level Interval ........................................................ 404 Service Level Maximizer ....................................................844 Service Level Maximizer Algorithm ....................................337 Service Level Supervisor ................................................... 404 Service Level Supervisor Call Selection Override ............. 337 Service Level Target (% in sec) ......................................... 405 Service Link Mode
Attendant Console ................................................ 71, 776 Service Objective .......................................................406, 994
Agent Login ID ..............................................................54 service observing ...............................................................333 Service Observing ..................................................... 123, 124 Service Observing (Basic) ................................................. 845 Service Observing (Remote/By FAC) ................................ 845 Service Observing (VDNs) .................................................845 Service Observing by Recording Device ........................... 133 Service Observing Listen/Talk Access Code ..................... 275 Service Observing Listen Only Access Code .................... 275
Service Observing Next Call Listen Only Access Code .....275 Service Observing No Talk Access Code .......................... 275 SERVICE OBSERVING PERMISSION ............................. 139 Service Type .............................. 500, 501, 512, 803, 804, 915 Session Connect Message Cntr ................................ 501, 804 Session Establishment Timer (min) ................................... 719 Session - Local/Remote .....................................................669 Session Manager
primary and secondary ............................................... 786 session refresh timer
non-SCCAN applications ............................................931 Set Color ............................................................................784 Set Layer 1 timer T1 to 30 seconds ...................................822 Sfx ......................................................................425, 547, 975 Shared UUI Feature Priorities ............................................540 Share Level ........................................................................795 sharing content ................................................................ 1016 Short/Prefixed Registration Allowed .......................... 477, 753 Short Holding Threshold .................................................... 951 Short Holding Time (seconds) ........................................... 951 Short Interdigit Timer ......................................................... 309 Show ANSWERED BY on Display ............................ 533, 952 Shuttle ................................................................................952 Side ............................................................................550, 553
DS1 Circuit Pack ........................................................ 236 Sig Bit Inversion .................................................................933 Sig Grp ...............................................................208, 547, 670 Signaling group ..................................................................700 Signaling Group ................................. 514, 585, 663, 916, 952 signaling group:Incoming Dialog Loopbacks ..................... 709 signaling group:RFC 3389 Comfort Noise ......................... 718 Signaling-group of Primary SM BW Mgr ............................486 Signaling-group of Secondary SM BW Mgr ....................... 486 Signaling Mode .......................................................... 237, 719 signaling trunks, Russian and R2-MFC ............................. 126 Signaling Type ................................................................... 754 signing up
PCNs and PSNs ....................................................... 1021 Silence Suppression .......................................................... 439 Simultaneous Calls ............................................................ 663 SIP
attendant .....................................................................739 console parameters ............................................ 163, 871 SIP 64K Data ..................................................................... 443 SIP Endpoint Managed Transfer ........................................362 SIP Reference Trunk Group .............................................. 538 SIP Signaling Group .......................................................... 719 SIP URI ..............................................................397, 786, 994 Site Data .................................................................... 681, 732 SIT Ineffective Other .......................................................... 729 sit no circuit ........................................................................730 SIT Reorder ....................................................................... 730 SIT Treatment for Call Classification ................................. 728 SIT Vacant Code ................................................................731 Size Enhanced List .............................................28, 30, 32, 34 Personal List ...............................................28, 30, 32, 34
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Index
Size (continued) System List ................................................. 28, 30, 32, 34
Skill Hunt Group ................................................................. 406
SL Agent Login ID ..............................................................54
Slip Detection .....................................................................240 SLM Count Abandoned Calls ............................................ 406 Slot
IP Interfaces ............................................................... 452 SN
Agent Login ID ..............................................................54 SNMP notification .............................................................. 183
emergency call ........................................................... 183 Softkey Labels ................................................................... 560 Software Package ..............................................................825 sort documents by last updated .......................................1016 Source ............................................................................... 628
Announcements Audio Sources ................................... 59 SOURCE ADDRESSES .................................................... 484 Source File .........................................................................368 Source No ..........................................................................628 SPDU Cntr ................................................................. 502, 804 Speaker ............................................................................. 784 Speakerphone ................................................................... 754 speakerphone group paging .............................................. 370 Special Application ............................................................ 867 Special Character for Restricted Number .......................... 187 Special Dialing Option ....................................................... 756 SPECIAL DIALING OPTION ............................................. 195
Attendant Console ........................................................ 73 Special Dial Tone ....................................................... 309, 350 Special Dial Tone for Digital/IP Stations ............................ 350 Special Digit Conversion ....................................................866 Speed ........................................................................ 494, 596
IP Interfaces ............................................................... 455 SPEEDS ............................................................................ 199 SPID .......................................................................... 200, 555 SPID -- (Service Profile Identifier) .................................... 756 split supervisor ...................................................................273 SPRT Retransmissions ......................................................446 SPRT TC0 Payload Size ....................................................446 SPRT TC1 Payload Size ....................................................446 SPRT TC1 Window Size ....................................................446 SPRT TC2 Payload Size ....................................................447 SPRT TC2 Window Size ....................................................447 SPRT TC3 Payload Size ....................................................447 SPRT Timer TC1-TA01 ...................................................... 447 SPRT Timer TC1-TA02 ...................................................... 447 SPRT Timer TC1-TR03 ..................................................... 447 SPRT Timer TC2-TA01 ...................................................... 448 SPRT Timer TC2-TA02 ...................................................... 448 SPRT Timer TC2-TR03 ..................................................... 448 src rgn ................................................................................472 SSE Inter-Packet Interval .................................................. 448 SSE Repetition Redundancy (Packets) ............................. 448 Stand alone cluster type .................................................... 586
Standard Factory Number ............................................... 1010 Start ................................................................................... 988 Start/End
Service Hours Table ................................................... 699 Start B Signal .....................................................................952 Start Date/Time ..................................................................368 Start Day ............................................................................375 Start Days (Sun through Sat)
Administered Connection ............................................. 42 Start Hour .......................................................................... 376 Starting Extension ..............................................................165 Starting Port
Access Endpoint ...........................................................38 Start Min ............................................................................ 376 Start Month ........................................................................ 376 Start Position ..................................................................... 953 Start Time
Administered Connection ............................................. 42 Station
Change Station Extension .......................................... 119 Station and Trunk MSP ......................................................840 Station as Virtual Extension ............................................... 840 Station-Button Display of UUI IE Data ............................... 133 Station Call Transfer Recall Timer ..................................... 309 Station Coverage Path For Coverage After Forwarding
System Parameters Call Coverage Call Forwarding .. 816 station extension ................................................................790 Station Extension ...............................................................792 Station Firmware Download Access Code ........................ 268 Station Hunt Before Coverage ...........................................813 Station Lock ....................................................................... 134 Station Lock Activation/Deactivation ..................................269 Station Lock COR .............................................................. 134 station name .......................................................................117 Station Name
Change Station Extension .......................................... 119 Station Putting Call On-Hold ..............................................360 Station Security Code Change .......................................... 270 Station Security Code verification ......................................697 Stations With Off PBX Telephone Integration .................... 794 stations with Off-PBX telephone integration ...................... 790 Stations With Off-PBX Telephone Integration
Dual Mode .................................................................. 791 Stations With System-wide Retrieval Permission .............. 291 Station Tone Forward Disconnect ......................................327 Station User Admin of FBI Assign ..................................... 270 Station User Button Ring Control .......................................270 Station When Call is Active ................................................360 Status .................................................................................503 status functions ..................................................................142 Status Poll VDN
Best Routing Service .................................................... 85 Step ................................................................................... 897 Stop Date/Time .................................................................. 368 Store to dsk ........................................................................803 Store VDN Name in Station's Local Call Log .....................332 Stub Network Region .........................................................460
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Index
Subnet Bits ........................................................................ 434 Subnet Mask ..............................................453, 456, 493, 496 Subnet Trunking ...................................................................63 Subscriber Number ............................................................670 Suite Check-in ................................................................... 387 Supervisor Extension .........................................................407 Supplementary Service Protocol ............................... 521, 933 Supplementary Services with Rerouting ............................849 Supply CPE Loopback Jack Power ................................... 241 support .............................................................................1018 Supported Set Type
Alias Station ..................................................................55 Support Request History ....................................................982 Suppress # Outpulsing .............................................. 533, 953 Suppress CDR for Ineffective Call Attempts ...................... 112 Survivable COR ................................................................. 755 Survivable GK Node Name ................................................755 Survivable Processor .........................................................798 Survivable Trunk Dest ........................................................755 SVN Authorization Code Violation Notification Enabled .... 694 SVN Login ..........................................................................694 SVN Remote Access Violation Notification Enabled ..........694 SVN Station Security Code Violation Notification Enabled 696 Switchhook Flash ...............................................................776 Switch Hook Query Response Timeout ............................. 346 Switch Name ......................................................................302 Switch Node .........................................................................85 Sync ...................................................................................472 Synchronization ................................................. 522, 596, 934 Synchronization over IP .....................................................361 Synch Source .................................................................... 551 System CESID Default ...................................................... 103 System ID .......................................................................... 757 System In Day Service ...................................................... 591 System In Night Service .................................................... 591 System List
Abbreviated Dialing ...................................................... 32 System Management Data Transfer .................................. 840 System Manager ................................................................692 System parameters
multifrequency signaling ............................................. 601 Security violation notification .............................. 693, 695 System parameters, mode code ........................................590 System parameters call coverage/ call forwarding: page 1 ....................................................................................807, 814 System Parameters Media Gateway Automatic Recovery Rule ................................................................................... 854 System parameters OCM call classification ...................... 859 System Parameters - SCCAN ........................................... 865 System Preferred ...............................................................801 SYSTEM PRINTER PARAMETERS ..................................300 System Updates Time On Station Displays ....................... 328
T
T2 (Backward Signal) Activation Timer (secs) ................... 355 T303 Timer (sec .................................................................719
T303 Timer (sec) ................................................................237 T3 Timer Length (sec) ........................................................551 Table Active ....................................................................... 891 TAC .................................................................... 514, 643, 916 Take Down Link for Lost Messages ................................... 381 Talk Duration ......................................................................731 Target % .............................................................................655 Target Socket Load ............................................................453 Target socket load and Warning level ................................ 453 TCP Signaling Link Establishment for Avaya H.323 Endpoints ............................................................................................ 469 TDD/TTY Mode ..................................................................444 Team Btn Display Name .................................................... 137 Team Btn Silent if Active .................................................... 137 Team Pick Up by Going Off Hook ...................................... 138 TEI ............................................................................. 200, 551
Station ........................................................................ 757 Telcordia Technologies NI-2 Operator Service Access ......689 Telecommuting Access ...................................................... 868 Telecommuting Access Extension ..................................... 868 Telephone Event Payload Type ......................................... 982 Temporary Bridged Appearance on Call Pickup ................ 365 Tenant ................................................................ 869, 872, 874
attendant group .......................................................... 869 Tenant Description ............................................................. 872 tenant number ......................................................................51 Tenant Partitioning ............................................................. 840 Terminal Dial-Up Access Code .......................................... 270 Terminal Endpoint Identifier (TEI) ...................................... 200 Terminal Equipment Identifier (TEI) ................................... 200 Terminal Parameters ..........................................................875 Terminal Trans. Init. (TTI) ...................................................841 Terminal Translation Initialization Enabled .........................293 Terminal Translation Initialization Merge Code .................. 270 Terminal Translation Initialization Separation Code ...........271 Terminate to Coverage Pts. with Bridged Appearances .... 178 Terminating Extension Group ............................................ 880 Termination Type ............................................................... 551 TestCall BCC ..................................................................... 514 Testcall ITC ........................................................................ 514 Testcall Service ..................................................................515 Tests
Station ........................................................................ 757 Text lines on Bulletin Board ................................................. 86 TFTP Server ...................................................................... 884 TFTP Server Node Name .................................................. 885 TFTP Server Port ...............................................................885 Threshold for Blocking Off-Net Redirection of Incoming Trunk Calls ................................................................................... 811 Tie Call ...............................................................................169 Time
system parameters port networks .............................. 864 Time (sec) to Drop Call on No Answer .............................. 953 Time (Start) ........................................................................ 206 Time (Stop) ........................................................................ 207 Time Before Off-Hook Alert ................................................298 Timed ACW ........................................................................845
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Index
Timed ACW Interval (sec) ..................................................407 Time Delay .........................................................................597 Time Format on Terminals
Location Parameters .................................................. 572 Time In Queue Warning (sec) ............................................ 166 Time Interval ...................................................................... 695 Time of Day Chart ..............................................................129 Time of Day Coverage Table ..................................... 886, 887 Time of Day Lock Table ..................................................... 757 Time of Day Routing .......................................................... 841 Time of Day Routing Plan .......................................... 888, 889 Time of Day Station Lock Table ......................................... 890 Time of Scheduled Emergency Access Summary Report . 386 Time of Scheduled Wakeup Activity Report .......................386 Time of Scheduled Wakeup Summary Report ...................386 Time Remaining .................................................................425 Time Reminder on Hold (sec) ............................................ 166 Time to Login ..................................................................... 247 Time Warning Extension .................................................... 397 Time Warning Threshold ....................................................398 Timezone Offset .................................................................567 TN .................... 62, 68, 72, 398, 563, 664, 677, 757, 883, 994
Agent LoginID ...............................................................51 Data Module ............................................................... 193 Group Paging Using Speakerphone ...........................372 TN2185 circuit pack ........................................................... 548 TN2198 circuit packs ......................................................... 552 TN2501 VAL Boards .......................................................... 827 TN2501 VAL Maximum Capacity ....................................... 841 TN2602 Boards with 320 VoIP Channels .......................... 827 TN2602 Boards with 80 VoIP Channels ............................ 827 TN556B circuit packs .........................................................552 to board Integrated Announcement Translations ......................427 To IP Address .................................................................... 434 toll ........................................................................................ 63 Toll Analysis ....................................................................... 892 Location ...................................................................... 893 Toll List .......................................................................692, 894 Toll Restricted .................................................................... 917 Tone (Frequency/Level) ..................................................... 897 Tone (msec) ............................................................... 540, 965 Tone Continuous ................................................................861 Tone Detection Mode System parameters country options ........................... 822 Tone Generation ................................................................ 896 Tone Generation Plan ........................................................ 572 Tone Name ................................................................ 861, 900 Total Administered ............................................................. 634 Total Administered Members ............................. 418, 547, 975 Total CPN Len ....................................................................638 Total Len ............................................................................ 634 Total Length ............................................................... 213, 797 CAMA Numbering Format .......................................... 103 Totals ................................................................................. 656 Touch Tone Sidetone (dB) ......................................... 875, 878 Touch Tone Transmit (dB) ..................................................878
training ............................................................................. 1017 Transfer ..............................................................................689 Transfer Conference .......................................................... 560 Transfer Into QSIG Voice Mail ........................................... 849 Transferred Ring Pattern ................................................... 328 Transfer to Voice Mail Access Code .................................. 271 Transfer Upon Hang-Up .....................................................314 Translation-ID Number Mismatch Interval (days) .............. 696 transmission mode for inter-system IP DTMF ................... 481 transmission mode for Intra-System IP DTMF ...................482 Transmit LBO .....................................................................242 Transmit Line Build-Out ..................................................... 242 Transmitted Signal Gain (dB) .............................................608 Transport Method ...............................................................720 Trk Grp(s) ...................................................................634, 638 Trk-to-Trk Restriction Override
Class Of Service .........................................................144 Truncate Station Number in ANI ........................................ 611 trunk access codes .............................................................. 68 Trunk Alerting Tone Interval ............................................... 310 Trunk Answer Any Station Access Code ........................... 271 Trunk Contact .................................................................... 968 Trunk COR .........................................................................310 Trunk Direction ...................................................................649 Trunk Flash ........................................................................ 917 Trunk Gain ................................................................. 649, 935 Trunk Group .......................................................................901
Format ........................................................................ 525 Trunk Group for Channel Selection ................................... 720 Trunk Groups ................................................................... 1010 Trunk Grp No ..................................................................... 797 Trunk Hunt ................................................................. 522, 935 Trunk Length ......................................................................968 Trunk Port .......................................................................... 650 trunk selection ....................................................................790 Trunk Selection
Stations With Off-PBX Telephone Integration .............792 Trunk Signaling Type ......................................................... 917 Trunk Termination ...................................................... 651, 935 Trunk-to-Trunk Transfer ..................................................... 289 Trunk Type ......................................................... 522, 650, 936 Trunk Type (in/out) .............................................................918 Trunk Vendor ..................................................................... 968 TSC
Route Pattern ............................................................. 692 TSC Index ..................................................................209, 670 TSC Method for Auto Callback .......................................... 538 TSC per MWI Interrogation ................................................416 TSC Supplementary Service Protocol ............................... 721 TTI Security Code ..............................................................293 TTI State ............................................................................ 293 TTL Contact ....................................................................... 968 TTL Type ............................................................................968 TTL Vendor ........................................................................ 969 Two-Digit Aux Work Reason Codes ...........................187, 346 two observers .................................................................... 333 Type ................................................................. 656, 975, 1010
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Index
Type (continued) Attendant Console ........................................................ 68 Call Type Analysis Table ...............................................90 CTI Link ...................................................................... 185 Data Module ............................................................... 193 Date and Time ............................................................ 203 Feature-Related System Parameters ......................... 289 IP Interfaces ............................................................... 454 Station ........................................................................ 758 Survivable Processor ..................................................801
Type (Column) ................................................................... 628 Type (field) ......................................................................... 628 Type of 3PCC Enabled ...................................................... 787
U
UCID Network Node ID ......................................................303 UDP-ARS Calls Considered Offnet ....................................216 UDP Extension Search Order ............................................217 UDP Port Range ................................................................ 462 UDP Port Range Max ........................................................ 462 UDP Port Range Min ......................................................... 462 Unacceptable Threshold
-+Dev - 2804 Hz ......................................................... 971 -+Dev - 404 Hz ........................................................... 971 --Dev - 2804 Hz .......................................................... 971 --Dev - 404 Hz ............................................................ 971 Max - 1004 Hz Loss ....................................................972 Maximum C Message Noise .......................................972 Maximum C Notched Noise ........................................972 Min - 1004 Hz Loss .....................................................972 Minimum ERL ............................................................. 972 Minimum SRL-HI ........................................................ 972 Minimum SRL-LO ....................................................... 972 Unanswered DID Call Timer .............................................. 310 Unauthorized Precedence Level ........................................622 Unhold ............................................................................... 314 Unicode ................................................................................66 Unicode Name ................................................................... 936 Agent Login ID ..............................................................51 Station login ................................................................763 VDN ............................................................................ 994 Uniform Dialing Plan .......................................................... 841 Uniform Dial Plan Table ..................................................... 983 Universal Call ID ........................................................ 303, 542 Unknown Numbers Considered Internal for AUDIX ...........319 Unnamed Registrations and PSA for IP Telephones ......... 294 Unrestricted Call List ..................................................129, 895 Upon DTE LOS ..................................................................242 Upper Bound (msec) ..........................................................574 USA Default Algorithm .......................................................860 Usage Alloc ........................................................................516 Usage Allocation Enhancements .......................................841 USA SIT Algorithm .............................................................860 Use COR for All Group II Responses ................................ 608 Use COR for Calling Party Category ................................. 608 Used
Used (continued) System Parameters Customer Options page 1 .......... 825 System Parameters Customer Options page 2 .......... 827 Virtual MAC Addresses .............................................1012
Use Default Server Parameters .........................................466 Used for BSR Polling
Vector Directory Number .......................................... 1001 Used for DCS .....................................................................953 Used for SIP Stations ........................................................ 692 Used Only for Paging .........................................................954 Use Enhanced Formats ..................................................... 113 Use ISDN Layouts ..............................................................113 Use Legacy CDR Formats ................................................. 113 User Control Restrict Activation/Deactivation .................... 271 User Guidance Display ...................................................... 322 User Profile ........................................................................ 986 Use System Level Parameter Values ................................ 494 Use System Syslog Values ................................................ 497 Use Time Adjustments from Location ................................ 699 Use VDN Time Zone For Holiday Vectoring .................... 1007 USNI Calling Name for Outgoing Calls ..............................319 US NI Delayed Calling Name Update ................................533 UUI IE Treatment ............................................................... 534
V
V.32 ....................................................................................445 V.34 ....................................................................................445 V.42 Error Correction ......................................................... 448 V.90 ....................................................................................445 V.92 ....................................................................................446 VAC ....................................................................................989 Vacant Code ...................................................................... 622 Validate BCMS/VuStats Login IDs ..................................... 340 Value Added (VALU) .......................................................... 849 Var ............................................................................989, 1005 Variables for Vectors .......................................................... 987 VDN Name .................................................................542, 656 VDN of Origin Annc. Extension ..........................................995 VDN of Origin Announcement ................................... 127, 847 VDN Override for ASAI Messages ...................................1001 VDN Return Destination .................................................... 847 VDN Timed ACW Interval ................................................ 1002 VDN Time Zone Offset .....................................................1007 VDN Variables ..................................................................1005 Vector .................................................................................245
Hunt Group ................................................................. 398 Vector Directory Number ................................................... 990
Send VDN as Called Ringing Name Over QSIG ...... 1001 Vector Disconnect Timer (min) ...........................................328 Vectoring (ANI/II-Digits Routing) ........................................845 Vectoring (Basic) ................................................................845 Vectoring (Best Service Routing) .......................................845 Vectoring (CINFO) ............................................................. 846 Vectoring (G3V4 Advanced Routing) .................................846 Vectoring (Holidays) ...........................................................846 Vectoring (Prompting) ........................................................ 846
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Index
Vector Number ...................................................................995 Vector Variable x ................................................................ 276 Verify Wakeup Announcement Access Code .................... 278 Version ...............................................................................919 Video (Norm) ......................................................................473 Video (Prio) ........................................................................ 473 Video (Shr) .........................................................................473 Video 802.1p Priority ..........................................................463 Video Bridge .................................................................... 1008 video capable IP Softphones ............................................. 826 video capable stations ....................................................... 827 Video PHB Value ............................................................... 463 videos .............................................................................. 1018 View Buttons ...................................................................... 561 viewing
PCNs ........................................................................ 1020 PSNs ........................................................................ 1020 VIP Caller ...........................................................................147 VIP Wakeup ....................................................................... 386 VIP Wakeup Reminder Call ............................................... 169 VIP Wakeups Per 5 Minutes .............................................. 386 Virtual Channel Identifier ....................................................721 Virtual MAC Addresses ....................................................1012 Virtual Path Identifier ..........................................................721 VIS FEATURE OPTIONS .................................................... 71 VLAN ......................................................................... 434, 454 VMS Hunt Group Extension ...............................................592 VOA messages ..................................................................127 Voice Coverage Message Retrieval Access Code ............ 271 Voice Do Not Disturb Access Code ................................... 278 Voice Mail Extension ..........................................................416 Voice Mail Handle .............................................................. 417 Voice Mail Hunt Group Ext .................................................314 Voice mail Number .............................................................784 Voice Mail Number .....................................................209, 417 Voice Paging, COR ............................................................581 Voice Paging, TN ............................................................... 582 Voice Paging TAC Loudspeaker Paging ...................................................581 Voice Paging Timeout (sec) ....................................... 581, 954 Voice Principal Message Retrieval Access Code .............. 271 Voice Receive (dB) .................................................... 876, 878 Voice Sidetone (dB) ................................................... 876, 878 Voice Transmit (dB) ........................................... 876, 878, 879 Voip Channel License Limit ............................................... 585 Volume for DCP Types ...................................................... 877 Volume for IP Types ...........................................................877 Vustats ............................................................................... 561 VuStats ...................................................................... 340, 847 VuStats (G3V4 Enhanced) .................................................847 VuStats Objective .............................................................. 407
wakeup announcement ......................................................278 WAN-BW limits (Total) ....................................................... 473 WAN-BW-limits (Units) .......................................................474 warning tone on service observing .................................... 333 watch list .......................................................................... 1016 Whisper Page .................................................................... 357 Whisper Page Activation Access Code ............................. 272 Whisper Page Tone Given To ............................................ 357 Wideband Access Endpoint .................................................36 Wideband Support ..................................................... 534, 954 Wideband Support Options ........................................544, 664 Wideband Switching .......................................................... 842 Width ..................................................................................664 Wink Timer (msec) .............................................................937 Wireless ............................................................................. 842 Without Flash .....................................................................314 WNDP Emergency 911 Route String ................................. 625 WNDP PRECEDENCE ACCESS CODES ........................ 281 Worldwide Numbering Dial Plan Active ............................. 626
X
XID .............................................................................200, 763 XID Test ............................................................................. 555 XMOBILE Type .................................................................. 763 X-Mobility/Wireless Type ................................................... 722 XOIP Endpoint type ........................................................... 763 XOIP Treatment ................................................................. 938
Y
Year ....................................................................................203
Z
Zip Tone Burst For Callmaster Endpoints .......................... 344
W
WAE .....................................................................................36 Wait Answer Supervision Timer ......................................... 328 Wait time before failback (seconds) ...................................725
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