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Grandstream Networks, Inc.
UCM630X Series Enterprise-Grade Unified Communication Solutions User Manual
COPYRIGHT
©2021 Grandstream Networks, Inc. http://www.grandstream.com All rights reserved. Information in this document is subject to change without notice. Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted. The latest electronic version of this user manual is available for download here: http://www.grandstream.com/support Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc. in the United States, Europe, and other countries.
CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
WARNING
Please do not use a different power adaptor with your devices as it may cause damage to the products and void the manufacturer warranty.
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GNU GPL INFORMATION
UCM630X firmware contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: http://www.grandstream.com/support/faq/gnu-general-public-license/gnu-gpl-information-download
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Table of Content
DOCUMENT PURPOSE ............................................................................. 27
CHANGE LOG ............................................................................................ 28
Firmware Version 1.0.5.4............................................................................................................... 28 Firmware Version 1.0.3.10............................................................................................................. 28 Firmware Version 1.0.2.25............................................................................................................. 30
WELCOME.................................................................................................. 31
PRODUCT OVERVIEW .............................................................................. 32
Technical Specifications ................................................................................................................ 32
INSTALLATION........................................................................................... 36
Equipment Packaging.................................................................................................................... 36 Connect Your UCM630X (UCM6301 as example).......................................................................... 36
UCM6302 front and back view................................................................................................ 38 UCM6304 front and back view................................................................................................ 38 UCM6308 front and back view................................................................................................ 39
GETTING STARTED................................................................................... 40
Use the LCD Menu........................................................................................................................ 40 Use the LED Indicators.................................................................................................................. 42 Using the Web UI .......................................................................................................................... 43
Accessing the Web UI ............................................................................................................ 43 Setup Wizard ......................................................................................................................... 44 Main Settings ......................................................................................................................... 44 Web GUI Languages.............................................................................................................. 45 Web GUI Search Bar ............................................................................................................. 45 Saving and Applying Changes................................................................................................ 46 Setting Up an Extension ................................................................................................................ 46
SYSTEM SETTINGS................................................................................... 47
General Settings ........................................................................................................................... 47 HTTP Server ................................................................................................................................. 48 Network Settings ........................................................................................................................... 49
Basic Settings ........................................................................................................................ 49 802.1X ................................................................................................................................... 55
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Static Routes ......................................................................................................................... 56 Port Forwarding ..................................................................................................................... 59 ARP Settings.......................................................................................................................... 61 OpenVPN® ................................................................................................................................... 62 DDNS Settings .............................................................................................................................. 64 Security Settings ........................................................................................................................... 66 Static Defense ....................................................................................................................... 66 Dynamic Defense................................................................................................................... 70 Fail2ban................................................................................................................................. 71 SSH Access........................................................................................................................... 73 LDAP Server ................................................................................................................................. 74 LDAP Server Configurations................................................................................................... 75 LDAP Phonebook................................................................................................................... 77 LDAP Client Configurations .................................................................................................... 80 Time Settings ................................................................................................................................ 82 Automatic Date and Time....................................................................................................... 82 Set Date and Time ................................................................................................................. 83 NTP Server ............................................................................................................................ 84 Office Time ............................................................................................................................ 84 Holiday .................................................................................................................................. 85 Email Settings ............................................................................................................................... 87 Email settings ........................................................................................................................ 87 Email Templates..................................................................................................................... 89 Email Send Log...................................................................................................................... 90 TR-069.......................................................................................................................................... 93
PROVISIONING .......................................................................................... 94
Overview....................................................................................................................................... 94 Configuration Architecture for End Point Device............................................................................. 94 Auto Provisioning Settings............................................................................................................. 95 Discovery ...................................................................................................................................... 98 Uploading Devices List ................................................................................................................ 100 Managing Discovered Devices .................................................................................................... 100 Global Configuration.................................................................................................................... 101
Global Policy........................................................................................................................ 101 Global Templates ..................................................................................................................111 Model configuration ......................................................................................................................113 Model templates................................................................................................................... 113 Model Update ...................................................................................................................... 115 Device Configuration ....................................................................................................................116
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Create New Device .............................................................................................................. 117 Manage Devices .................................................................................................................. 117 Sample Application...................................................................................................................... 122
EXTENSIONS ........................................................................................... 126
Create New User......................................................................................................................... 126 Create New SIP Extension ................................................................................................... 126 Create New IAX Extension ................................................................................................... 136 Create New FXS Extension.................................................................................................. 143
Batch Add Extensions.................................................................................................................. 150 Batch Add SIP Extensions.................................................................................................... 150 Batch Add IAX Extensions.................................................................................................... 157
Batch Extension Resetting Functionality ...................................................................................... 162 Search and Edit Extension .......................................................................................................... 162 Export Extensions ....................................................................................................................... 163 Import Extensions........................................................................................................................ 164 Extension Details ........................................................................................................................ 171 E-mail Notification ....................................................................................................................... 172 Multiple Registrations per Extension ............................................................................................ 174 SMS Message Support................................................................................................................ 175
EXTENSION GROUPS ............................................................................. 176
Configure Extension Groups........................................................................................................ 176 Using Extension Groups.............................................................................................................. 177
ANALOG TRUNKS ................................................................................... 178
Analog Trunk Configuration ......................................................................................................... 178 PSTN Detection .......................................................................................................................... 182
VOIP TRUNKS .......................................................................................... 186
VoIP Trunk Configuration............................................................................................................. 186 Trunk Groups .............................................................................................................................. 198 Direct Outward Dialing (DOD)...................................................................................................... 199
SLA STATION ........................................................................................... 201
Create/Edit SLA Station............................................................................................................... 201 Sample Configuration .................................................................................................................. 202
CALL ROUTES ......................................................................................... 204
Outbound Routes ........................................................................................................................ 204
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Configuring Outbound Routes .............................................................................................. 204 Outbound Blacklist ............................................................................................................... 207 Scheduled Sync ................................................................................................................... 209 PIN Groups .......................................................................................................................... 209 Inbound Routes........................................................................................................................... 212 Inbound Rule Configurations ................................................................................................ 213 Inbound Route: Prepend Example........................................................................................ 217 Inbound Route: Multiple Mode.............................................................................................. 218 Inbound Route: Route-Level Mode ....................................................................................... 220 Inbound Route: Inbound Mode BLF Monitoring..................................................................... 221 Inbound Route: Import/Export Inbound Route....................................................................... 222 FAX with Two Media............................................................................................................. 223 Blacklist Configurations ........................................................................................................ 223
FAX SERVER ............................................................................................ 225
Configure Fax/T.38 ...................................................................................................................... 225 Receiving Fax ............................................................................................................................. 227
Example Configuration to Receive Fax from PSTN Line ....................................................... 227 Example Configuration for Fax-To-Email .............................................................................. 230 FAX Sending ............................................................................................................................... 232
AUDIO CONFERENCE............................................................................. 233
Conference Room Configurations................................................................................................ 233 Conference Call Operations ........................................................................................................ 236
Join a Conference Call ......................................................................................................... 236 Invite Other Parties to Join Conference ................................................................................ 236 During the Conference ......................................................................................................... 237 Google Service Settings Support.......................................................................................... 238 Conference Schedule.................................................................................................................. 241 Contact Group............................................................................................................................. 244 Conference Recordings............................................................................................................... 245 Conference Call Statistics............................................................................................................ 245
VIDEO CONFERENCE ............................................................................. 247
Video Conference........................................................................................................................ 247 Conference Schedule.................................................................................................................. 249 Wave WebRTC Video Calling & Conferencing ............................................................................. 251
IVR............................................................................................................. 254
Configure IVR ............................................................................................................................. 254
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Black/White List in IVR ................................................................................................................ 257 Create Custom Prompt................................................................................................................ 259
LANGUAGE SETTINGS FOR VOICE PROMPT ..................................... 261
Download and Install Voice Prompt Package ............................................................................... 261 Customize Specific Prompt.......................................................................................................... 263 Username Prompt Customization ................................................................................................ 263
Upload Username Prompt File from Web GUI ...................................................................... 263 Record Username via Voicemail Menu ................................................................................. 264
VOICEMAIL............................................................................................... 265
Configure Voicemail .................................................................................................................... 265 Access Voicemail ........................................................................................................................ 267 Leaving Voicemail ....................................................................................................................... 268 Voicemail Email Settings ............................................................................................................. 269 Configure Voicemail Group.......................................................................................................... 270
RING GROUP ........................................................................................... 272
Configure Ring Group.................................................................................................................. 272 Remote Extension in Ring Group................................................................................................. 275
PAGING AND INTERCOM GROUP ......................................................... 278
Configure Paging/Intercom Group ............................................................................................... 278 Configure Multicast Paging................................................................................................... 278 Configure 2-way Intercom .................................................................................................... 280 Configure 1-way Paging ....................................................................................................... 281 Configure Announcement Paging ......................................................................................... 283 Paging/Intercom Group Settings........................................................................................... 284
Configure a Scheduled Paging/Intercom...................................................................................... 284
CALL QUEUE ........................................................................................... 286
Configure Call Queue .................................................................................................................. 286 Call Center Settings and Enhancements...................................................................................... 291 Queue Statistics .......................................................................................................................... 292 Switchboard ................................................................................................................................ 296 Global Queue Settings ................................................................................................................ 299
PICKUP GROUPS .................................................................................... 301
Configure Pickup Groups............................................................................................................. 301 Configure Pickup Feature Code................................................................................................... 301
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MUSIC ON HOLD ..................................................................................... 303
BUSY CAMP-ON ...................................................................................... 306
PRESENCE............................................................................................... 307
FOLLOW ME............................................................................................. 310
SPEED DIAL ............................................................................................. 313
DISA .......................................................................................................... 314
EMERGENCY ........................................................................................... 316
CALLBACK............................................................................................... 320
BLF AND EVENT LIST ............................................................................. 321
BLF ............................................................................................................................................. 321 Event List .................................................................................................................................... 321
DIAL BY NAME......................................................................................... 324
Dial by Name Configuration......................................................................................................... 324
ACTIVE CALLS AND MONITOR ............................................................. 328
Active Calls Status ...................................................................................................................... 328 Hang Up Active Calls................................................................................................................... 330 Call Monitor................................................................................................................................. 330
CALL FEATURES..................................................................................... 332
Feature Codes ............................................................................................................................ 332 Parking Lot.................................................................................................................................. 337 Call Park ..................................................................................................................................... 339
Park a Call ........................................................................................................................... 339 Retrieve Parked Call ............................................................................................................ 339 Monitor Call Park CID Name Information (GXP21xx, GRP261x Phones Only) ...................... 339 Call Recording............................................................................................................................. 339 Enable Spy.................................................................................................................................. 340 Shared Call Appearance (SCA) ................................................................................................... 341
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ANNOUNCEMENT ................................................................................... 345
PBX SETTINGS ........................................................................................ 346
PBX Settings/General Settings .................................................................................................... 346 PBX Settings/RTP Settings.......................................................................................................... 349
RTP Settings........................................................................................................................ 349 Payload ............................................................................................................................... 350 PBX Settings/Voice Prompt Customization .................................................................................. 350 Record New Custom Prompt ................................................................................................ 350 Upload Custom Prompt ........................................................................................................ 351 Download All Custom Prompt............................................................................................... 352 PBX Settings/ Call Failure Tone Settings ..................................................................................... 352 SIP Trunk Prompt Tone ........................................................................................................ 352 General Call Prompt Tone .................................................................................................... 353 PBX Settings/Recordings Storage ............................................................................................... 354 PBX Settings/NAS....................................................................................................................... 356
SIP SETTINGS.......................................................................................... 358
SIP Settings/General ................................................................................................................... 358 SIP Settings/MISC....................................................................................................................... 358 SIP Settings/Session Timer ......................................................................................................... 359 SIP Settings/TCP and TLS .......................................................................................................... 360 SIP Settings/NAT......................................................................................................................... 361 SIP Settings/TOS ........................................................................................................................ 361 SIP Settings/STIR/SHAKEN ........................................................................................................ 363 Transparent Call-Info header ....................................................................................................... 365
IAX SETTINGS ......................................................................................... 366
IAX Settings/General ................................................................................................................... 366 IAX Settings/Registration............................................................................................................. 366 IAX Settings/Security................................................................................................................... 367
INTERFACE SETTINGS ........................................................................... 368
UCM RemoteConnect.............................................................................. 371
Plan Settings............................................................................................................................... 372 Custom logo................................................................................................................................ 373 Statistics ..................................................................................................................................... 374
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API CONFIGURATION ............................................................................. 375
API Configuration Parameters ..................................................................................................... 375 API Queries Supported ........................................................................................................ 375
Upload Voice Prompt via API....................................................................................................... 377
CTI SERVER ............................................................................................. 379
ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) ............ 380
CRM INTEGRATION................................................................................. 381
SugarCRM .................................................................................................................................. 381 VTigerCRM ................................................................................................................................. 382 ZohoCRM ................................................................................................................................... 384 Salesforce CRM .......................................................................................................................... 386 ACT! CRM................................................................................................................................... 387
PMS INTEGRATION ................................................................................. 389
HMobile PMS Connector ............................................................................................................. 389 HSC PMS ................................................................................................................................... 390 Mitel PMS ................................................................................................................................... 391 IDS PMS ..................................................................................................................................... 392 PMS API ..................................................................................................................................... 392 Connecting to PMS ..................................................................................................................... 393 PMS Features ............................................................................................................................. 394
Room Status ........................................................................................................................ 394 Wake Up Service ................................................................................................................. 395 Mini Bar ............................................................................................................................... 396
WAKEUP SERVICE .................................................................................. 399
Wake Up Service using Admin Login ........................................................................................... 399 Wake Up Service from User Portal .............................................................................................. 400 Wake Up Service using Feature Code ......................................................................................... 401
ANNOUNCEMENTS CENTER ................................................................. 402
Announcements Center Settings ................................................................................................. 403 Group Settings ............................................................................................................................ 403
STATUS AND REPORTING ..................................................................... 406
PBX Status.................................................................................................................................. 406
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Trunks.................................................................................................................................. 406 Extensions ........................................................................................................................... 407 Interfaces Status .................................................................................................................. 408 System Status............................................................................................................................. 409 General................................................................................................................................ 409 Network ............................................................................................................................... 410 Storage Usage ..................................................................................................................... 411 Resource Usage .................................................................................................................. 412 System Events ............................................................................................................................ 413 Alert Events List ................................................................................................................... 413 Alert Log .............................................................................................................................. 419 Alert Contact ........................................................................................................................ 420 CDR............................................................................................................................................ 421 Downloaded CDR File.......................................................................................................... 428 CDR Export Customization................................................................................................... 429 CDR in GDMS Cloud ........................................................................................................... 430 Statistics .............................................................................................................................. 430 Recording Files .................................................................................................................... 432
USER PORTAL ......................................................................................... 433
Basic Information ........................................................................................................................ 435 Personal Data ............................................................................................................................. 435 Value-added Features ................................................................................................................. 435
MAINTENANCE ........................................................................................ 436
User Management....................................................................................................................... 436 User Information .................................................................................................................. 436 Custom Privilege.................................................................................................................. 437 Concurrent Multi-User Login................................................................................................. 440 Change Password................................................................................................................ 441 Change Username ............................................................................................................... 442 Change binding Email .......................................................................................................... 442
Operation Log ............................................................................................................................. 444 Upgrading ................................................................................................................................... 446
No Local Firmware Servers .................................................................................................. 447 Backup........................................................................................................................................ 448
Backup/Restore ................................................................................................................... 448 Data Sync ............................................................................................................................ 450 Restore Configuration from Backup File ............................................................................... 452 System Cleanup/Reset................................................................................................................ 453
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Reset and Reboot ................................................................................................................ 453 Cleaner ................................................................................................................................ 453 USB/SD Card Files Cleanup ................................................................................................ 458 System Recovery ........................................................................................................................ 458 Syslog ......................................................................................................................................... 460 Network Troubleshooting ............................................................................................................. 461 Ethernet Capture.................................................................................................................. 461 IP Ping ................................................................................................................................. 463 Traceroute............................................................................................................................ 464 Signaling Troubleshooting ........................................................................................................... 464 Analog Record Trace ........................................................................................................... 464 Service Check ............................................................................................................................. 466 Network Status............................................................................................................................ 467
EXPERIENCING THE UCM630X SERIES IP PBX .................................. 468
Table of Tables
Table 1: Technical Specifications.................................................................................................................... 32 Table 2: UCM630X Equipment Packaging...................................................................................................... 36 Table 3: LCD Menu Options........................................................................................................................... 41 Table 4: UCM6304/6308 LED Indicators ........................................................................................................ 42 Table 5: General Settings Parameters............................................................................................................ 47 Table 6: HTTP Server Settings....................................................................................................................... 48 Table 7: UCM630X Network SettingsBasic Settings.................................................................................... 49 Table 8: UCM630X Network Settings802.1X............................................................................................... 56 Table 9: UCM630X Network SettingsStatic Routes ..................................................................................... 57 Table 10: UCM630X Network SettingsPort Forwarding ............................................................................... 59 Table 11: ARP Settings .................................................................................................................................. 62 Table 12: UCM630X System SettingsNetwork SettingsOpenVPN® ......................................................... 62 Table 13: UCM630X FirewallStatic DefenseCurrent Service .................................................................... 66 Table 14: Typical Firewall Settings ................................................................................................................. 67 Table 15: Firewall Rule Settings ..................................................................................................................... 69 Table 16: UCM630X Firewall Dynamic Defense ............................................................................................. 70 Table 17: Fail2Ban Settings ........................................................................................................................... 72 Table 18: SSH Access ................................................................................................................................... 74 Table 19: Time Auto Updating ........................................................................................................................ 83
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Table 20: Create New Office Time.................................................................................................................. 84 Table 21: Create New Holiday........................................................................................................................ 86 Table 22: Email Settings ................................................................................................................................ 87 Table 23: Email Log Display Filter ............................................................................................................... 91 Table 24: Email Codes................................................................................................................................... 92 Table 25: Auto Provision Settings................................................................................................................... 97 Table 26: Global Policy Parameters Localization ....................................................................................... 102 Table 27: Global Policy Parameters Phone Settings .................................................................................. 103 Table 28: Global Policy Parameters Contact List ....................................................................................... 104 Table 29: Global Policy Parameters Maintenance...................................................................................... 105 Table 30: Global Policy Parameters Network Settings ............................................................................... 107 Table 31: Global Policy Parameters Customization ................................................................................... 109 Table 32: Global Policy Parameters Communication Settings.................................................................... 110 Table 33: Create New Template ................................................................................................................... 111 Table 34: Create New Model Template......................................................................................................... 113 Table 35: SIP Extension Configuration ParametersBasic Settings............................................................. 127 Table 36: SIP Extension Configuration ParametersMedia ......................................................................... 129 Table 37: SIP Extension Configuration ParametersFeatures ..................................................................... 130 Table 38: SIP Extension Configuration ParametersSpecific Time .............................................................. 136 Table 39: Table 34: SIP Extension Configuration ParametersFollow Me .................................................... 136 Table 40: IAX Extension Configuration ParametersBasic Settings............................................................. 137 Table 41: IAX Extension Configuration ParametersMedia ......................................................................... 138 Table 42: IAX Extension Configuration ParametersFeatures ..................................................................... 139 Table 43: IAX Extension Configuration ParametersSpecific Time .............................................................. 142 Table 44: IAX Extension Configuration ParametersFollow Me................................................................... 142 Table 45: FXS Extension Configuration ParametersBasic Settings............................................................ 143 Table 46: FXS Extension Configuration ParametersMedia ........................................................................ 144 Table 47: FXS Extension Configuration ParametersFeatures .................................................................... 145 Table 48: FXS Extension Configuration ParametersSpecific Time............................................................. 149 Table 49: FXS Extension Configuration ParametersFollow Me.................................................................. 149 Table 50: Batch Add SIP Extension Parameters ........................................................................................... 150 able 51: Batch Add IAX Extension Parameters............................................................................................. 157 Table 52: SIP extensions Imported File Example.......................................................................................... 165 Table 53: IAX extensions Imported File Example.......................................................................................... 167 Table 54: FXS Extensions Imported File Example ........................................................................................ 169 Table 55: Analog Trunk Configuration Parameters........................................................................................ 178 Table 56: PSTN Detection for Analog Trunk ................................................................................................. 185 Table 57: Create New SIP Trunk .................................................................................................................. 186 Table 58: SIP Register Trunk Configuration Parameters............................................................................... 188 Table 59: SIP Peer Trunk Configuration Parameters .................................................................................... 192
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Table 60: Create New IAX Trunk .................................................................................................................. 195 Table 61: IAX Register Trunk Configuration Parameters............................................................................... 196 Table 62: IAX Peer Trunk Configuration Parameters .................................................................................... 197 Table 63: SLA Station Configuration Parameters.......................................................................................... 201 Table 64: Outbound Route Configuration Parameters .................................................................................. 204 Table 65: Outbound Routes/Scheduled Sync ............................................................................................... 209 Table 66: Outbound Routes/PIN Group ........................................................................................................ 209 Table 67: Inbound Rule Configuration Parameters ....................................................................................... 213 Table 68: FAX/T.38 Settings......................................................................................................................... 226 Table 69: Conference Room Configuration Parameters................................................................................ 233 Table 70: Conference Settings ..................................................................................................................... 235 Table 71: Conference Caller IVR Menu ........................................................................................................ 237 Table 72: Conference Schedule Parameters ................................................................................................ 241 Table 73: Video Conference room Configuration Parameters ....................................................................... 247 Table 74: Conference Settings ..................................................................................................................... 248 Table 75: Video Conference Schedule Parameters ...................................................................................... 249 Table 76: IVR Configuration Parameters ...................................................................................................... 255 Table 77: Voicemail Settings ........................................................................................................................ 266 Table 78: Voicemail IVR Menu ..................................................................................................................... 267 Table 79: Voicemail Email Settings .............................................................................................................. 269 Table 80: Voicemail Group Settings.............................................................................................................. 271 Table 81: Ring Group Parameters................................................................................................................ 272 Table 82: Multicast Paging Configuration Parameters .................................................................................. 279 Table 83: 2-way Intercom Configuration Parameters .................................................................................... 280 Table 84: 1-way Paging Configuration Parameters ....................................................................................... 282 Table 85: Announcement Paging Configuration Parameters ......................................................................... 283 Table 86: Schedule Paging / Intercom Settings ............................................................................................ 285 Table 87: Call Queue Configuration Parameters........................................................................................... 286 Table 88: Static Agent Limitation .................................................................................................................. 290 Table 89: Call Center Parameters ................................................................................................................ 291 Table 90: Switchboard Parameters .............................................................................................................. 297 Table 91: Global Queue Settings.................................................................................................................. 300 Table 92: SIP Presence Status..................................................................................................................... 308 Table 93: Follow Me Settings ....................................................................................................................... 311 Table 94: Follow Me Options........................................................................................................................ 312 Table 95: DISA Settings ............................................................................................................................... 315 Table 96: Emergency Numbers Parameters ................................................................................................. 318 Table 97: Callback Configuration Parameters............................................................................................... 320 Table 98: Event List Settings........................................................................................................................ 321 Table 99: UCM630X Feature Codes............................................................................................................. 332
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Table 100 : Parking Lot ................................................................................................................................ 338 Table 101: Add SCA Private Number............................................................................................................ 343 Table 102: Editing the SCA Number ............................................................................................................. 344 Table 103: Announcement Parameters ........................................................................................................ 345 Table 104: Internal Options/General............................................................................................................. 346 Table 105: Internal Options/RTP Settings..................................................................................................... 349 Table 106: Internal Options/Payload............................................................................................................. 350 Table 107: NAS Settings .............................................................................................................................. 356 Table 108: SIP Settings/General .................................................................................................................. 358 Table 109: SIP Settings/Misc ....................................................................................................................... 358 Table 110: SIP Settings/Session Timer......................................................................................................... 359 Table 111: SIP Settings/TCP and TLS .......................................................................................................... 360 Table 112: SIP Settings/NAT ........................................................................................................................ 361 Table 113: SIP Settings/ToS......................................................................................................................... 361 Table 114: SIP Settings/STIR/SHAKEN - Add Authentication Number Settings ............................................. 363 Table 115: SIP Settings/STIR/SHAKEN Certificate Settings....................................................................... 364 Table 116: IAX Settings/General .................................................................................................................. 366 Table 117: IAX Settings/Registration ............................................................................................................ 366 Table 118: IAX Settings/Static Defense ........................................................................................................ 367 Table 119: PBX Interface Settings................................................................................................................ 369 Table 120: Configuration Parameters (New)................................................................................................. 375 Table 121: New API Supported Queries ....................................................................................................... 375 Table 122: API Configuration Parameters..................................................................................................... 377 Table 123: SugarCRM Settings.................................................................................................................... 381 Table 124: vTigerCRM Settings.................................................................................................................... 383 Table 125: ZohoCRM Settings ..................................................................................................................... 385 Table 126: Salesforce Settings..................................................................................................................... 386 Table 127: PMS Supported Features ........................................................................................................... 389 Table 128: PMS Basic Settings .................................................................................................................... 393 Table 129: PMS Wake up Service ................................................................................................................ 395 Table 130: Create New Mini Bar................................................................................................................... 396 Table 131: Create New Maid........................................................................................................................ 397 Table 132: Wakeup Service ......................................................................................................................... 400 Table 133: Max Wakeup Members ............................................................................................................... 400 Table 134: Announcements Center Settings................................................................................................. 403 Table 135: Group Settings ........................................................................................................................... 403 Table 136: Trunk Status ............................................................................................................................... 407 Table 137: Extension Status......................................................................................................................... 408 Table 138: Interface Status Indicators .......................................................................................................... 409 Table 139: System StatusGeneral ............................................................................................................ 410
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Table 140: System StatusNetwork ............................................................................................................ 410 Table 141: Alert Events ................................................................................................................................ 413 Table 142: Alert Contact............................................................................................................................... 420 Table 143: CDR Filter Criteria ...................................................................................................................... 422 Table 144: CDR Statistics Filter Criteria ....................................................................................................... 431 Table 145: User ManagementCreate New User ........................................................................................ 437 Table 146: Change Binding Email option...................................................................................................... 442 Table 147: Operation Log Column Header ................................................................................................... 445 Table 148: Data Sync Configuration ............................................................................................................. 451 Table 149: Automatic Cleaning Configuration ............................................................................................... 455 Table 150: USB/SD Card Files Cleanup ....................................................................................................... 458 Table 151: Ethernet Capture ........................................................................................................................ 462
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Table of Figures
Figure 1: UCM6301 Top View ........................................................................................................................ 36 Figure 2: UCM6301 Back View ...................................................................................................................... 37 Figure 3: UCM6301 Front View...................................................................................................................... 37 Figure 4: UCM6302 Back View ...................................................................................................................... 38 Figure 5:UCM6302 Front View....................................................................................................................... 38 Figure 6: UCM6304 Front View...................................................................................................................... 38 Figure 7: UCM6304 Back View ...................................................................................................................... 38 Figure 8: UCM6308 Front View...................................................................................................................... 39 Figure 9: UCM6308 Back View ...................................................................................................................... 39 Figure 10: Ports Status .................................................................................................................................. 42 Figure 11: UCM6302 Web GUI Login Page.................................................................................................... 43 Figure 12: UCM630X Setup Wizard ............................................................................................................... 44 Figure 13: UCM630X Web GUI Language ..................................................................................................... 45 Figure 14: Web GUI Search Bar .................................................................................................................... 46 Figure 15: General Settings Interface............................................................................................................. 47 Figure 16: UCM6302 Network Interface Method: Route ................................................................................. 53 Figure 17: UCM6302 Network Interface Method: Switch ................................................................................ 54 Figure 18: UCM6302 Network Interface Method: Dual.................................................................................... 55 Figure 19: UCM630X Using 802.1X as Client................................................................................................. 55 Figure 20: UCM630X Using 802.1X EAP-MD5............................................................................................... 56 Figure 21: UCM6304 Static Route Sample..................................................................................................... 58 Figure 22: UCM6304 Static Route Configuration............................................................................................ 59 Figure 23: Create New Port Forwarding ......................................................................................................... 60 Figure 24: UCM630X Port Forwarding Configuration...................................................................................... 61 Figure 25: GXP2160 Web Access using UCM6302 Port Forwarding .............................................................. 61 Figure 26: Open VPN® Feature on the UCM630X ......................................................................................... 63 Figure 27: Register Domain Name on noip.com ............................................................................................. 64 Figure 28: UCM630X DDNS Setting .............................................................................................................. 65 Figure 29: Using Domain Name to Connect to UCM630X .............................................................................. 65 Figure 30: Create New Firewall Rule.............................................................................................................. 68 Figure 31: Configure Dynamic Defense.......................................................................................................... 71 Figure 32: Fail2ban Settings .......................................................................................................................... 72 Figure 33: SSH Access.................................................................................................................................. 74 Figure 34: LDAP Server Configurations ......................................................................................................... 75 Figure 35: Default LDAP Phonebook DN ....................................................................................................... 76 Figure 36: Default LDAP Phonebook Attributes .............................................................................................. 76 Figure 37: LDAP ServerLDAP Phonebook.................................................................................................. 77 Figure 38: Add LDAP Phonebook .................................................................................................................. 77
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Figure 39: Edit LDAP Phonebook .................................................................................................................. 78 Figure 40: Import Phonebook......................................................................................................................... 78 Figure 41: Phonebook CSV File Format......................................................................................................... 78 Figure 42: LDAP Phonebook After Import....................................................................................................... 79 Figure 43: Export Selected LDAP Phonebook ................................................................................................ 80 Figure 44: LDAP Client Configurations........................................................................................................... 80 Figure 45: GXP2170 LDAP Phonebook Configuration.................................................................................... 82 Figure 46: Set Time Manually ........................................................................................................................ 83 Figure 47: Create New Office Time ................................................................................................................ 84 Figure 48: SettingsTime SettingsOffice Time ........................................................................................... 85 Figure 49: Create New Holiday ...................................................................................................................... 86 Figure 50: SettingsTime SettingsHoliday ................................................................................................. 87 Figure 51: UCM630X Email Settings.............................................................................................................. 89 Figure 52: Email Template ............................................................................................................................. 90 Figure 53: Email Send Log ............................................................................................................................ 91 Figure 54: Email Logs.................................................................................................................................... 92 Figure 55: Zero Config Configuration Architecture for End Point Device ......................................................... 95 Figure 56: UCM630X Zero Config.................................................................................................................. 96 Figure 57: Auto Provision Settings ................................................................................................................. 97 Figure 58: Auto Discover................................................................................................................................ 99 Figure 59: Discovered Devices ...................................................................................................................... 99 Figure 60: Device List - CSV file Sample...................................................................................................... 100 Figure 61: Managing Discovered Devices .................................................................................................... 100 Figure 62: Global Policy Categories ............................................................................................................. 102 Figure 63: Edit Global Template ................................................................................................................... 112 Figure 64: Edit Model Template.................................................................................................................... 114 Figure 65: OEM Models............................................................................................................................... 115 Figure 66: Template Management................................................................................................................ 116 Figure 67: Upload Model Template Manually................................................................................................ 116 Figure 68: Create New Device ..................................................................................................................... 117 Figure 69: Manage Devices ......................................................................................................................... 118 Figure 70: Edit Device ................................................................................................................................. 118 Figure 71: Edit Customize Device Settings................................................................................................... 120 Figure 72: Modify Selected Devices - Same Model ...................................................................................... 121 Figure 73: Modify Selected Devices - Different Models................................................................................. 121 Figure 74: Device List in Zero Config ........................................................................................................... 122 Figure 75: Zero Config Sample - Global Policy............................................................................................. 123 Figure 76: Zero Config Sample - Device Preview 1 ...................................................................................... 124 Figure 77: Zero Config Sample - Device Preview 2 ...................................................................................... 124 Figure 78: Zero Config Sample - Device Preview 3 ...................................................................................... 125
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Figure 79: Create New Device ..................................................................................................................... 126 Figure 80: Manage Extensions .................................................................................................................... 162 Figure 81: Export Extensions ....................................................................................................................... 164 Figure 82: Import Extensions ....................................................................................................................... 164 Figure 83: Import File................................................................................................................................... 165 Figure 84: Import Error ................................................................................................................................ 171 Figure 85: Extension Details ........................................................................................................................ 172 Figure 86: E-mail Notification - Prompt Information ...................................................................................... 172 Figure 87: Account Registration Information................................................................................................. 173 Figure 88: GSWave Settings and QR Code ................................................................................................. 173 Figure 89: Multiple Registrations per Extension............................................................................................ 174 Figure 90: Extension - Concurrent Registration ............................................................................................ 174 Figure 91: SMS Message Support ............................................................................................................... 175 Figure 92: Edit Extension Group .................................................................................................................. 176 Figure 93: Select Extension Group in Outbound Route ................................................................................ 177 Figure 94: UCM630X FXO Tone Settings..................................................................................................... 182 Figure 95: UCM630X PSTN Detection ......................................................................................................... 183 Figure 96: UCM630X PSTN Detection: Auto Detect ..................................................................................... 184 Figure 97: UCM630X PSTN Detection: Semi-Auto Detect ............................................................................ 184 Figure 98: Trunk Group................................................................................................................................ 198 Figure 99: Trunk Group Configuration .......................................................................................................... 199 Figure 100: DOD extension selection........................................................................................................... 200 Figure 101: Edit DOD .................................................................................................................................. 200 Figure 102: SLA Station............................................................................................................................... 201 Figure 103: Enable SLA Mode for Analog Trunk ........................................................................................... 202 Figure 104: Analog Trunk with SLA Mode Enabled ....................................................................................... 202 Figure 105: SLA Example - SLA Station....................................................................................................... 202 Figure 106: SLA Example - MPK Configuration............................................................................................ 203 Figure 107: Country Codes.......................................................................................................................... 208 Figure 108: Blacklist Import/Export .............................................................................................................. 208 Figure 109: Create New PIN Group ............................................................................................................. 210 Figure 110: PIN Members ............................................................................................................................ 210 Figure 111: Outbound PIN............................................................................................................................ 210 Figure 112: CDR Record.............................................................................................................................. 211 Figure 113: Importing PIN Groups from CSV files......................................................................................... 211 Figure 114: Incorrect CSV File ..................................................................................................................... 211 Figure 115: CSV File Format........................................................................................................................ 212 Figure 116: CSV File Successful Upload...................................................................................................... 212 Figure 117: Inbound Route feature: Prepend................................................................................................ 218 Figure 118: Inbound Route - Multiple Mode.................................................................................................. 219
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Figure 119: Inbound Route - Multiple Mode Feature Codes .......................................................................... 220 Figure 120: Inbound Route - Route-Level Mode........................................................................................... 220 Figure 121: Global Inbound Mode................................................................................................................ 221 Figure 122: Inbound Mode - Default Mode ................................................................................................... 222 Figure 123: Inbound Mode - Mode 1 ............................................................................................................ 222 Figure 124: Import/Export Inbound Route .................................................................................................... 222 Figure 125: Blacklist Configuration Parameters............................................................................................ 224 Figure 126: Blacklist csv File........................................................................................................................ 224 Figure 127: Fax Settings.............................................................................................................................. 225 Figure 128: Configure Analog Trunk............................................................................................................. 228 Figure 129: Configure Extension for Fax Machine: FXS Extension ............................................................... 228 Figure 130: Configure Extension for Fax Machine: Analog Settings.............................................................. 229 Figure 131: Configure Inbound Rule for Fax................................................................................................. 229 Figure 132: Create Fax Extension................................................................................................................ 230 Figure 133: Inbound Route to Fax Extension ............................................................................................... 231 Figure 134: List of Fax Files......................................................................................................................... 231 Figure 135: Fax Sending in Web GUI........................................................................................................... 232 Figure 136: Fax Send Progress ................................................................................................................... 232 Figure 137: Conference ............................................................................................................................... 235 Figure 138: Conference Invitation from Web GUI ......................................................................................... 236 Figure 139: Google Service SettingsOAuth2.0 Authentication................................................................... 239 Figure 140: Google ServiceNew Project ................................................................................................... 239 Figure 141: Google ServiceCreate New Credential................................................................................... 240 Figure 142: Google ServiceOAuth2.0 Login.............................................................................................. 240 Figure 143: Conference Schedule................................................................................................................ 243 Figure 144: Conference Scheduled-Ongoing ............................................................................................... 243 Figure 145: Conference Schedule-Completed.............................................................................................. 244 Figure 146: Contact Group .......................................................................................................................... 244 Figure 147: Conference Recordings............................................................................................................. 245 Figure 148: Conference Call Statistics ......................................................................................................... 245 Figure 149: Conference Report on Web....................................................................................................... 246 Figure 150: Conference Report on CSV....................................................................................................... 246 Figure 151: Video Conference Schedule...................................................................................................... 250 Figure 152: Video Conference Scheduled-Ongoing...................................................................................... 250 Figure 153: Video Conference Scheduled-Completed.................................................................................. 251 Figure 154: Enabling WebRTC Feature ....................................................................................................... 251 Figure 155: Enabling WebRTC on Extensions.............................................................................................. 252 Figure 156: Grandstream Wave Interface..................................................................................................... 252 Figure 157: Download Wave Client .............................................................................................................. 253 Figure 158: Create New IVR ........................................................................................................................ 254
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Figure 159: Key Pressing Events ................................................................................................................. 257 Figure 160: Black/Whitelist .......................................................................................................................... 259 Figure 161: Click on Prompt to Create IVR Prompt ...................................................................................... 260 Figure 162: Language Settings for Voice Prompt ......................................................................................... 261 Figure 163: Voice Prompt Package List........................................................................................................ 262 Figure 164: New Voice Prompt Language Added ......................................................................................... 262 Figure 165: Upload Single Voice Prompt for Entire Language Pack.............................................................. 263 Figure 166: Voicemail Settings..................................................................................................................... 265 Figure 167: Voicemail Email Settings ........................................................................................................... 270 Figure 168: Voicemail Group........................................................................................................................ 270 Figure 169: Ring Group ............................................................................................................................... 272 Figure 170: Ring Group Configuration.......................................................................................................... 275 Figure 171: Sync LDAP Server option.......................................................................................................... 276 Figure 172: Manually Sync LDAP Server ..................................................................................................... 276 Figure 173: Ring Group Remote Extension .................................................................................................. 277 Figure 174: Multicast Paging........................................................................................................................ 278 Figure 175: 2-way Intercom ......................................................................................................................... 280 Figure 176: 1-way Paging ............................................................................................................................ 281 Figure 177: Announcement Paging .............................................................................................................. 283 Figure 178: Page/Intercom Group Settings .................................................................................................. 284 Figure 179: Schedule Paging/Intercom page................................................................................................ 284 Figure 180: Creating a scheduled paging/intercom call ................................................................................ 285 Figure 181: Call Queue................................................................................................................................ 286 Figure 182: Agent Login Settings ................................................................................................................. 290 Figure 183: Call Queue Statistics................................................................................................................. 293 Figure 184: Queue's call log details ............................................................................................................. 293 Figure 185: Automatic Download Settings - Queue Statistics........................................................................ 294 Figure 186: Agent details ............................................................................................................................. 295 Figure 187: Login Record ............................................................................................................................ 295 Figure 188: Pause Log ................................................................................................................................ 296 Figure 189: Switchboard Summary .............................................................................................................. 296 Figure 190: Call Queue Switchboard............................................................................................................ 297 Figure 191: Queue Chairman....................................................................................................................... 298 Figure 192: Queue Agent............................................................................................................................. 299 Figure 193: Global Queue Settings .............................................................................................................. 299 Figure 194: Edit Pickup Group ..................................................................................................................... 301 Figure 195: Edit Pickup Feature Code ......................................................................................................... 302 Figure 196: Music On Hold Default Class..................................................................................................... 303 Figure 197: Play Custom Prompt ................................................................................................................. 304 Figure 198: Information Prompt ................................................................................................................... 305
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Figure 199: Record Custom Prompt............................................................................................................. 305 Figure 200: SIP Presence Configuration ...................................................................................................... 307 Figure 201: SIP Presence Feature Code...................................................................................................... 308 Figure 202: Presence Status CDR ............................................................................................................... 309 Figure 203: Edit Follow Me .......................................................................................................................... 310 Figure 204: Speed Dial Destinations ............................................................................................................ 313 Figure 205: List of Speed Dial...................................................................................................................... 313 Figure 206: Create New DISA...................................................................................................................... 314 Figure 207: Emergency Number Configuration............................................................................................. 317 Figure 208: 911 Emergency Sample ............................................................................................................ 319 Figure 209: Create New Event List .............................................................................................................. 322 Figure 210: Create Dial by Name Group ...................................................................................................... 324 Figure 211: Configure Extension First Name and Last Name........................................................................ 325 Figure 212: Dial By Name Group In IVR Key Pressing Events...................................................................... 326 Figure 213: Dial By Name Group In Inbound Rule........................................................................................ 327 Figure 214: StatusPBX StatusActive Calls - Ringing.............................................................................. 328 Figure 215: StatusPBX StatusActive Calls Call Established................................................................ 328 Figure 216: Call Connection less than half hour ........................................................................................... 329 Figure 217: Call Connection between half an hour and one hour.................................................................. 329 Figure 218: Call Connection more than one hour ......................................................................................... 330 Figure 219: Configure to Monitor an Active Call............................................................................................ 330 Figure 220: Enable/Disable Feature codes................................................................................................... 337 Figure 221: Parking Lot ............................................................................................................................... 337 Figure 222: New Parking Lot........................................................................................................................ 338 Figure 223: Monitored Call Park CID name .................................................................................................. 339 Figure 224: Download Recording File from CDR Page................................................................................. 340 Figure 225: Download Recording File from Recording Files Page ................................................................ 340 Figure 226: Enabling SCA option under Extension's Settings ....................................................................... 341 Figure 227: SCA Number Configuration ....................................................................................................... 342 Figure 228: SCA Private Number Configuration ........................................................................................... 342 Figure 229: SCA Options ............................................................................................................................. 343 Figure 230: Announcement settings ............................................................................................................. 345 Figure 231: Record New Custom Prompt..................................................................................................... 351 Figure 232: Upload Custom Prompt ............................................................................................................. 352 Figure 233: Download All Custom Prompt .................................................................................................... 352 Figure 234: SIP Trunk Prompt Tone ............................................................................................................. 353 Figure 235: General call Failure Prompts ..................................................................................................... 354 Figure 236: SettingsRecordings Storage .................................................................................................. 354 Figure 237: Recordings Storage Prompt Information.................................................................................... 355 Figure 238: Recording Storage Category ..................................................................................................... 356
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Figure 239:SIP Settings/STIR/SHAKEN - Add Authentication Number.......................................................... 363 Figure 240: SIP Settings/STIR/SHAKEN Certificate Settings..................................................................... 364 Figure 241: Transparent Call-Info................................................................................................................. 365 Figure 242: FXS Ports Signaling Preference ................................................................................................ 368 Figure 243: FXO Ports ACIM Settings .......................................................................................................... 368 Figure 244: DAHDI Settings......................................................................................................................... 370 Figure 245: RemoteConnect ........................................................................................................................ 371 Figure 246: UCM RemoteConnect - Effective Plan....................................................................................... 372 Figure 247: UCM RemoteConnect Plan Settings.......................................................................................... 373 Figure 248: Custom Logo ............................................................................................................................ 373 Figure 249: Concurrent Remote Calls .......................................................................................................... 374 Figure 250: Upload Prompt User Configuration ............................................................................................ 378 Figure 251: CTI Server Listening port .......................................................................................................... 379 Figure 252: SugarCRM Basic Settings......................................................................................................... 381 Figure 253: CRM User Settings ................................................................................................................... 382 Figure 254: vTigerCRM Basic Settings......................................................................................................... 383 Figure 255: CRM User Settings ................................................................................................................... 384 Figure 256: ZohoCRM Basic Settings .......................................................................................................... 384 Figure 257: CRM User Settings ................................................................................................................... 385 Figure 258: Salesforce Basic Settings.......................................................................................................... 386 Figure 259: Salesforce User Settings........................................................................................................... 387 Figure 260: Enabling ACT! CRM .................................................................................................................. 387 Figure 261: Enabling CRM on the User Portal.............................................................................................. 388 Figure 262: UCM & PMS interaction ............................................................................................................ 390 Figure 263: UCM & HSC PMS interaction .................................................................................................... 391 Figure 264: UCM & Mitel PMS interaction .................................................................................................... 392 Figure 265: UCM & IDS PMS interaction...................................................................................................... 392 Figure 266: Create New Room .................................................................................................................... 394 Figure 267: Room Status ............................................................................................................................. 394 Figure 268: Add batch rooms ....................................................................................................................... 395 Figure 269: Create New Wake Up Service ................................................................................................... 395 Figure 270: Wakeup Call executed .............................................................................................................. 396 Figure 271: Create New Mini Bar ................................................................................................................. 396 Figure 272: Create New Maid ...................................................................................................................... 397 Figure 273: Create New Consumer Goods................................................................................................... 397 Figure 274: Mini Bar .................................................................................................................................... 398 Figure 275: Create New Wakeup Service..................................................................................................... 399 Figure 276: Announcements Center............................................................................................................. 402 Figure 277: Announcements Center Group Configuration............................................................................. 404 Figure 278: Announcements Center Code Configuration.............................................................................. 405
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Figure 279: Announcements Center Example .............................................................................................. 405 Figure 280: StatusPBX Status .................................................................................................................. 406 Figure 281: Trunk Status.............................................................................................................................. 406 Figure 282: Extension Status ....................................................................................................................... 407 Figure 283: UCM6304 Interfaces Status ...................................................................................................... 408 Figure 284: System StatusStorage Usage ................................................................................................ 412 Figure 285: System StatusResource Usage ............................................................................................. 413 Figure 286: Alert Event List.......................................................................................................................... 414 Figure 287: System EventsAlert Events Lists: Disk Usage ........................................................................ 415 Figure 288: System EventsAlert Events Lists: External Disk Usage .......................................................... 415 Figure 289: System EventsAlert Events Lists: Memory Usage .................................................................. 416 Figure 290: System EventsAlert Events Lists: System Crash .................................................................... 416 Figure 291:System EventsAlert Events Lists: Register SIP Failed............................................................. 417 Figure 292: System EventsAlert Events Lists: Register SIP Trunk Failed .................................................. 417 Figure 293: System EventsAlert Log......................................................................................................... 419 Figure 294: Filter for Alert Log...................................................................................................................... 420 Figure 295: CDR Filter................................................................................................................................. 422 Figure 296: Call Report................................................................................................................................ 425 Figure 297: Call Report Entry with Audio Recording File .............................................................................. 426 Figure 298: Automatic Download Settings.................................................................................................... 427 Figure 299: CDR Report .............................................................................................................................. 427 Figure 300: Detailed CDR Information ......................................................................................................... 428 Figure 301: Downloaded CDR File Sample .................................................................................................. 428 Figure 302: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ...................................... 428 Figure 303: Downloaded CDR File Sample - Source Channel and Dest Channel 2 ...................................... 429 Figure 304: CDR Export File data ................................................................................................................ 430 Figure 305: CDR in GDMS Cloud ................................................................................................................ 430 Figure 306: CDR Statistics........................................................................................................................... 431 Figure 307: CDRRecording Files .............................................................................................................. 432 Figure 308: Edit User Information by Super Admin....................................................................................... 433 Figure 309: User Portal Login ...................................................................................................................... 434 Figure 310: User Portal Layout .................................................................................................................... 434 Figure 311: User Management Page Display ............................................................................................... 436 Figure 312: Create New User ...................................................................................................................... 436 Figure 313: User Management New Users................................................................................................ 437 Figure 314: Assign Backup permission to "Admin" users .............................................................................. 438 Figure 315: General User ............................................................................................................................ 439 Figure 316: Create New Custom Privilege ................................................................................................... 440 Figure 317: Multiple User Operation Error Prompt........................................................................................ 441 Figure 318: Change Password..................................................................................................................... 441
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Figure 319: Change Username.................................................................................................................... 442 Figure 320: Change Binding Email............................................................................................................... 442 Figure 321: Login Timeout Settings.............................................................................................................. 444 Figure 322: Operation Logs ......................................................................................................................... 445 Figure 323: Operation Logs Filter................................................................................................................. 446 Figure 324: Local Upgrade .......................................................................................................................... 446 Figure 325: Upgrading Firmware Files ......................................................................................................... 446 Figure 326: Reboot UCM630X..................................................................................................................... 447 Figure 327: Create New Backup .................................................................................................................. 448 Figure 328: Restore Warning ....................................................................................................................... 449 Figure 329: Backup / Restore ...................................................................................................................... 449 Figure 330: Local Backup ............................................................................................................................ 450 Figure 331: Data Sync ................................................................................................................................. 451 Figure 332: Restore UCM630X from Backup File......................................................................................... 452 Figure 333: Reset and Reboot ..................................................................................................................... 453 Figure 334: Manual Cleaning ....................................................................................................................... 454 Figure 335: Automatic Cleaning ................................................................................................................... 455 Figure 336: USB/SD Card Files Cleanup ..................................................................................................... 458 Figure 337: UCM6302 Recovery Web Page................................................................................................. 459 Figure 338: Recovery Mode......................................................................................................................... 460 Figure 339: Ethernet Capture....................................................................................................................... 462 Figure 340: Ping .......................................................................................................................................... 463 Figure 341: Traceroute ................................................................................................................................ 464 Figure 342: Troubleshooting Analog Trunks ................................................................................................. 465 Figure 343: A Key Dial-up FXO .................................................................................................................... 466 Figure 344: Service Check........................................................................................................................... 466 Figure 345: Network Status ......................................................................................................................... 467
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DOCUMENT PURPOSE
The intent of this document is to provide device administrators an overview of the specifications and features of the Grandstream UCM630X IPPBX system. To learn more about the UCM630X, please visit http://www.grandstream.com/support to download additional guides.
This guide covers following main topics:
Product overview Installation Getting started System settings Provisioning Extensions Extension groups Analog Trunks VoIP Trunks SLA station Call routes Conference room. Conference schedule IVR Language settings for voice prompt Voicemail Ring group Paging and intercom group Call queue Pickup groups PIN Groups Music on hold Fax Server
Busy camp-on Presence Follow me Speed Dial DISA Callback BLF and event list Dial by name Active calls and monitor Call features Call recording CTI Server Asterisk manager interface (AMI) CRM integration PMS integration Wakeup service Announcements center Status and reporting CDR (Call Details Record) User Portal Upgrading and maintenance Backup/restore Troubleshooting
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CHANGE LOG
This section documents significant changes from previous versions of the UCM630X user manuals. Only major new features or major document updates are listed here. Minor updates for corrections or editing are not documented here.
Firmware Version 1.0.5.4
Added support for v-Fax, Fax-sending, Email2Fax. [FAX SERVER] Added ability to restore backups remotely stored on GDMS [Restore Configuration from Backup File] Added support for STIR/SHAKEN. [SIP Settings/STIR/SHAKEN] [VoIP Trunk Configuration] Added ability to restrict calls and features based on CPU usage and data partition usage. [General
Settings] Added NAT option to the Export File Data filtering option. [Export File Data] Wave desktop client is now supported. [Wave WebRTC Video Calling & Conferencing] Added support for IAX. [IAX SETTINGS] [EXTENSIONS] [VoIP Trunk Configuration] Improved audio and video conference pages. A list of the meetings that have not started and a meeting
history list have been added. [Conference Schedule] [Conference Schedule]. Support sending post-meeting reports to the host after scheduling a meeting. [Conference Schedule]
[Conference Schedule]. Added the SRTP Debugging option to the Ethernet Capture page. [Enable SRTP Debugging] Updated the System Cleanup/Reset page interface. [System Cleanup/Reset] Added manual cleaning. [Cleaner] Added ability to add a custom browser tab icon and custom logos on various pages of the web
management portal and Wave Web portal. [Custom logo] Improved push notification for WAVE mobile. Added Instant Messaging functionality to Wave Web. [Wave WebRTC Video Calling & Conferencing] LDAP phonebook information will now be synchronized when viewing the Contacts page. [Wave WebRTC
Video Calling & Conferencing] UCM can now synchronize system event alerts to GDMS. [Plan Settings] Added threshold-based Call Control & Data Write Control. [General Settings] Added Layer 3 QoS for SIP and Layer 3 QoS for RTP options to global policy and relevant templates.
[Layer 3 QoS For RTP] [Layer 3 QoS For SIP] Improved device list import support and added the ability to export devices in the ZeroConfig Device List
page. [Managing Discovered Devices]
Firmware Version 1.0.3.10
Added RemoteConnect Mode option in SIP trunk. [RemoteConnect Mode]
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Cloud Storage for CDR Backup and Record. [CDR in GDMS Cloud] Add support for IDS PMS interface. [IDS PMS] Add ability to enable PMS Wakeup call from remote extensions. [PMS Remote Wakeup Service] Add ability to configure local country code for outbound route. [Local Country Code] Add support for Pattern and Leading digit filtering options in Callee Number. [Callee Number] Added description of SSH switch options in LCD menu. [Use the LCD Menu] Add the ability to import/export DOD. [Direct Outward Dialing (DOD)] Add the ability to enable auto record per inbound/outbound route. [Auto Record] [Auto Record] Add the ability to Enable Recording Whitelist. [Enable Recording Whitelist] Add the ability to select missed call type to be sent via email. [Missed Call Type] The Filter action is now supported in AMI sessions. [ASTERISK MANAGER INTERFACE] Added command to delete call recordings after downloading them. [recapi] Added commands to add, edit, and delete PIN groups. [addPinSets] [getPinSets] [updatePinSets]
[deletePinSets] Added External Disk Status alert event for monitoring external storage connection status. [Alert Events
List] The language of column titles in exported CDR reports and statistics reports will now be based on the
UCM's display language. [Downloaded CDR File] Added Parking Lot Timeout Alert-Info option. This will add the specified alert-info header value to parking
timeout callbacks. [Parking Lot Timeout Alert-Info] Updated HMobile check-in request format. [HMobile PMS Connector] Updated HMobile Mini Bar request. [HMobile PMS Connector] Added several updates to HSC PMS. [HSC PMS] Added the ability to specify the reason for agent pause (*83 by default). [Agent Pause] Added CDR Stored in GDMS Cloud option for RemoteConnect Plan Settings. [UCM RemoteConnect] GDMS Cloud Storage has been added as a recording storage location in the PBX. [PBX
Settings/Recordings Storage] After adding a UCM to GDMS, its RemoteConnect address will automatically be added as a SIP server
to the GDMS account. [UCM RemoteConnect] Added the Ringback Tone option. Users can now select a custom prompt to play as ringback tone for
callers dialing in via the selected inbound route. [Ringback tone] Ring Group Voicemail can now be set as a routing destination and an IVR key press destination. [Default
Destination] Added the following custom privileges: LDAP Server, UCM RemoteConnect, and Announcement.
[Custom Privilege] Added a Forgot Password option to the Wave Web login page. [Wave WebRTC Video Calling &
Conferencing] Added a new command to set same-day wakeup service by dialing *36. [Wakeup Service]
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Firmware Version 1.0.2.25
This is the initial version.
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WELCOME
Thank you for purchasing Grandstream UCM630X series IP PBX appliance. The UCM6300 series allows businesses to build powerful and scalable unified communication and collaboration solutions. This series of IP PBXs provide a platform that unifies all business communication on one centralized network, including voice, video calling, video conferencing, video surveillance, web meetings, data, analytics, mobility, facility access, intercoms and more. The UCM6300 series supports up to 3000 users and includes a built-in web meetings and video conferencing solution that allows employees to connect from the desktop, mobile, GVC series devices and IP phones. It can be paired with the UCM6300 ecosystem to offer a hybrid platform that combines the control of an on-premises IP PBX with the remote access of a cloud solution. The UCM6300 ecosystem consists of the Wave app for desktop and mobile, which provides a hub for collaborating remotely, and UCM RemoteConnect, a cloud NAT traversal service for ensuring secure remote connections. The UCM6300 series also offers cloud setup and management through GDMS and an API for integration with third-party platforms. By offering a highend unified communications and collaboration solution packed with a suite of mobility, security, meeting and collaboration tools, the UCM6300 series provides a powerful platform for any organization. The UCM6301/6302 support 1 x ARM Cortex-A53 Quad-Core CPU and 2GB of memory while UCM6304/6308 support 2 x ARM Cortex-A53 Quad-Core CPU and 2GB for master and 1GB slave memory.
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Caution: Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
Warning: Please do not use a different power adaptor with the UCM630X as it may cause damage to the product and void the manufacturer warranty. --------------------------------------------------------------------------------------------------------------------------------------------
This document is subject to change without notice. The latest electronic version of this user manual is available for download here: http://www.grandstream.com/support
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.
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PRODUCT OVERVIEW
Technical Specifications
The following table resumes all the technical specifications including the protocols / standards supported, voice codecs, telephony features, languages, and upgrade/provisioning settings for UCM630X series.
Interfaces
Analog Telephone FXS Ports
PSTN Line FXO Ports
Network Interfaces NAT Router Peripheral Ports
LED Indicators
LCD Display Reset Switch Voice/Video Capabilities Voice-over-Packet Capabilities Voice and Fax Codecs
Table 1: Technical Specifications
UCM6301: 1 port with lifeline support UCM6302: 2 ports with lifeline support UCM6304: 4 ports with lifeline support UCM6308: 8 ports with lifeline support Each port supports 3 REN UCM6301: 1 port UCM6302: 2 ports UCM6304: 4 ports UCM6308: 8 ports All ports have lifeline capability in case of power outage Three self-adaptive Gigabit ports (switched , routed or dual card mode) with PoE+ Yes (supports router mode and switch mode) UCM6301: USB 3.0, and SD card interface UCM6302: USB 2.0, USB 3.0, and SD card interface UCM6304/6308: 2x USB 3.0 and SD card interface UCM6301/UCM6302: None UCM6304/6308: Power 1/2, FXS, FXO, LAN, WAN, Heartbeat UCM6301/UCM6302: 320*240 LCD with touch screen for Shortcut Keys and
Scroll Bar UCM6304/6308: 128x32 dot matrix graphic LCD with DOWN and OK buttons Yes, long press for factory reset and short press for reboot
LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection & auto-switch to G.711, NetEQ, FEC 2.0, jitter resilience up to 50% audio packet loss Opus, G.711 A-law/U-law, G.722, G722.1 G722.1C, G.723.1 5.3K/6.3K, G.726-32, G.729A/B, iLBC, GSM; T.38
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Video Codecs
H.264, H.263, H263+, H.265, VP8
QoS
Layer 2 QoS (802.1Q, 802.1p) and Layer 3 (ToS, DiffServ, MPLS) QoS
Signaling and Control
DTMF Methods
Inband, RFC4733, and SIP INFO
Provisioning Protocol and Plug-and-Play
Mass provisioning using AES encrypted XML configuration file, auto-discovery & auto-provisioning of Grandstream IP endpoints via ZeroConfig (DHCP Option 66 multicast SIP SUBSCRIBE mDNS), eventlist between local and remote trunk
Network Protocols
TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH, HTTP/HTTPS, PPPoE, STUN, SRTP, TLS, LDAP, HDLC, HDLC-ETH, PPP, Frame Relay (pending), IPv6, OpenVPN®
API
Full API available for third-party platform and application integration
Disconnect Methods
Busy/Congestion/Howl Tone, Polarity Reversal, Hook Flash Timing, Loop Current Disconnect
Security
Media Encryption
SRTP, TLS1.2, HTTPS, SSH, 802.1x
Physical
Universal Power Supply
UCM6301/6302: Input: 100 ~ 240VAC, 50/60Hz; Output: DC+12V, 1.5A UCM6304/6308: Input: 100~240VAC,50/60Hz; Output: DC+12V, 2A
Dimensions
UCM6301/6302: 270mm(L) x 175mm(W) x 36mm(H) UCM6304/6308: 485mm(L) x 187.2mm(W) x 46.2mm(H)
Environmental Mounting
Operating: 32 - 113ºF / 0 ~ 45ºC, Humidity 10 - 90% (non-condensing) Storage: 14 - 140ºF / -10 ~ 60ºC, Humidity 10 - 90% (non-condensing) Wall mount (sold separately) & Desktop for UCM6301/6302 and Desktop & Rack mount for UCM6304/6308.
Weight
UCM6301: Unit weight 715g, Package weight 1211g UCM6302: Unit weight 725g, Package weight 1221g UCM6304: Unit weight 2490g, Package weight 3260g UCM6308: Unit weight 2550g, Package weight 3320g
Additional Features
Multi-language Support Caller ID
Web UI: English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, German, Russian, Italian, Polish, Czech, Turkish
Customizable IVR/voice prompts: English, Chinese, British English, German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese, Russian, Swedish, Turkish, Hebrew, Arabic, Netherlands
Customizable language pack to support any other languages Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 BT, NTT
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Polarity Reversal/ Wink Call Center Customizable Auto Attendant Telephony Operating System
Maximum Call Capacity
Maximum Attendees of Conference Bridges
Yes, with enable/disable option upon call establishment and termination Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability/workload, in-queue announcement
Up to 5 layers of IVR (Interactive Voice Response) in multiple languages
Based on Asterisk version 16
UCM6301 UCM6302 UCM6304 UCM6308 UCM6301 UCM6302 UCM6304 UCM6308
Users: 500 Concurrent calls (G.711): 75 Max concurrent SRTP calls (G.711): 50 Users: 1000 Concurrent calls (G.711): 150 Max concurrent SRTP calls (G.711): 100 Users: 2000 Concurrent calls (G.711): 300 Max concurrent SRTP calls (G.711): 200 Users: 3000 Concurrent calls (G.711): 450 Max concurrent SRTP calls (G.711): 300
4 Video Conference rooms and up to 20 parties with 1080p HD H.264 and Opus (assuming 4 video feeds + 1 screen sharing)
Voice Conference: Up to 75 parties with G.711 6 Video Conference rooms and up to 30 parties with
1080p HD H.264 and Opus (assuming 4 video feeds + 1 screen sharing) Voice Conference: Up to 150 parties with G.711 8 Video Conference rooms and up to 60 parties with 1080p HD H.264 and Opus (assuming 4 video feeds + 1 screen sharing) Voice Conference: Up to 200 parties with G.711 10 Video Conference rooms and up to 80 parties with 1080p HD H.264 and Opus (assuming 4 video feeds + 1 screen sharing) Voice Conference: Up to 300 parties with G.711
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Call Features Wave Mobile App Firmware Upgrade Compliance
Call park, call forward, call transfer, call waiting, caller ID, call record, call history, ringtone, IVR, music on hold, call routes, DID, DOD, DND, DISA, ring group, ring simultaneously, time schedule, PIN groups, call queue, pickup group, paging/intercom, voicemail, call wakeup, SCA, BLF, voicemail to email, speed dial, call back, dial by name, emergency call, call follow me, blacklist/whitelist, voice conference, video conference, eventlist, feature codes, busy camp-on/ call completion, voice control
Allows Android & iOS users to join UCM-hosted meetings & communicate with other users/solutions registered to the UCM6300
Supported by Grandstream Device Management System (GDMS), a zero-touch cloud provisioning and management system, It provides a centralized interface to provision, manage, monitor, and troubleshoot Grandstream products
FCC: Part 15 (CFR 47) Class B, Part 68 CE: EN 55032, EN 55035, EN61000-3-2, EN61000-3-3, EN 62368.1, ES 203
021, ITU K.21 IC: ICES-003, CS-03 Part I Issue 9 RCM: AS/NZS CISPR 32, AS/NZS 62368.1, AS/CA S002, AS/CA S003.1/.2 Power adapter: UL 60950-1 or UL 62368-1
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Note: UCM630X FXS ports lifeline functionality: The UCM630X FXS interfaces are metallic through to the FXO interfaces. If there is power outage, FXS1 port will fail over to FXO 1 port, FXS 2 port will fail over to FXO 2 port. The user can still access the PSTN connected with the FXO interfaces from FXS interfaces. ---------------------------------------------------------------------------------------------------------------------------------------------------
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INSTALLATION
Before deploying and configuring the UCM630X series, the device needs to be properly powered up and connected to a network. This section describes detailed information on installation, connection, and warranty policy of the UCM630X series.
Equipment Packaging
Table 2: UCM630X Equipment Packaging
Main Case
1
Power Adaptor
1
Ethernet Cable
1
Quick Installation Guide
1
Connect Your UCM630X (UCM6301 as example)
Interface status Information
Settings
Back Home Figure 1: UCM6301 Top View
Navigation Keys OK
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WAN port Heartbeat port FXO port
Reset LAN port FXS port
Figure 2: UCM6301 Back View
Ground
SD card slot USB port
Figure 3: UCM6301 Front View
To set up the UCM6301, follow the steps below:
1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6301. 2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub. 3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6301. Insert the
main plug of the power adapter into a surge-protected power outlet. 4. Wait for the UCM6301 to boot up. The LCD in the front will show the device hardware information when the
boot process is done. 5. Once the UCM6301 is successfully connected to network, press the Home button to display the IP address. 6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax)
to the FXS ports.
Note:
The ground screw needs to be connected.
The same steps will be used in order to connect UCM6302/6304/6308
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UCM6302 front and back view
WAN port Heartbeat port 2x FXO ports
Reset LAN port 2x FXS ports
Figure 4: UCM6302 Back View
Ground
Figure 5:UCM6302 Front View
SD card slot 2x USB ports
UCM6304 front and back view
Navigation keys
Power LED 4x FXS LEDs LAN LED
2x USB ports
OK button
4x FXO LEDs
WAN LED
Heartbeat LED
Figure 6: UCM6304 Front View
Power WAN port Heartbeat port
4x FXO ports
SD card slot
Ground
LAN port
4x FXS ports Figure 7: UCM6304 Back View
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UCM6308 front and back view
Navigation keys
Power LED 8x FXS LEDs LAN LED
2x USB ports
OK button 8x FXO LEDs
WAN LED Heartbeat LED
SD card slot
Figure 8: UCM6308 Front View
Power WAN port Heartbeat port
8x FXO ports
Ground
LAN port
8x FXS ports Figure 9: UCM6308 Back View
Safety Compliances
The UCM630X series IP PBX complies with FCC/CE and various safety standards. The UCM630X power adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM630X package only. The manufacturer's warranty does not cover damages to the device caused by unsupported power adapters.
Warranty
If the UCM630X series IP PBX was purchased from a reseller, please contact the company where the device was purchased for replacement, repair, or refund. If the device was purchased directly from Grandstream, contact our Technical Support Team for an RMA (Return Materials Authorization) number before the product is returned. Grandstream reserves the right to remedy warranty policy without prior notification.
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Warning: Use the power adapter provided with the UCM630X series IP PBX. Do not use a different power adapter as this may damage the device. This type of damage is not covered under warranty. ---------------------------------------------------------------------------------------------------------------------------------------------------
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GETTING STARTED
To get started with the UCM630X setup process, use the following available interfaces: LCD display, and web portal.
The LCD display shows hardware, software, interface status and network information and can be navigated via the Slide control and Touch keys. From here, users can configure basic network settings, run diagnostic tests, and factory reset.
The web portal (may also be referred to as web UI in this guide) is the primary method of configuring the UCM.
This section will provide step-by-step instructions on how to use these interfaces to quickly set up the UCM and start making and receiving calls with it.
Use the LCD Menu
Idle Screen Once the device has booted up completely, the LCD will show the UCM model, hardware version and IP address. Upon menu key press timeout (30 seconds), the screen will default back to this information.
Menu Pressing the Home button will show the main menu. All available menu options are found in [Table 3: LCD Menu Options].
Menu Navigation Scrolling sown using slide control through the menu options. Press the OK button to select an option.
Exit Selecting the Back option will return to the previous menu. For the Device Info, Network Info, and Web Info screens that have no Back option, pressing the OK button will return to the previous menu.
LCD Backlight The LCD backlight will turn on upon button press and will go off when idle for 30 seconds.
The following table summarizes the layout of the LCD menu of UCM630x.
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View Events Device Info Network Info Network Menu
Factory Menu
Table 3: LCD Menu Options
Critical Events Other Events
Hardware: Hardware version number Software: Software version number P/N: Part number WAN MAC: WAN side MAC address LAN MAC: LAN side MAC address Uptime: System uptime
WAN Mode: DHCP, Static IP, or PPPoE WAN IP: IP address WAN Subnet Mask LAN IP: IP address LAN Subnet Mask
WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE Static Route Reset: Select this to reset static route settings.
Reboot Factory Reset LCD Test Patterns
Press DOWN and OK buttons to scroll through and select different LCD patterns to test. Once a test is done, press the OK button to return to the previous menu.
Fan Mode Select Auto or On.
LED Test Patterns All On, All Off, and Blinking are the available options. Selecting Back in the menu will revert the LED indicators back to their actual status.
RTC Test Patterns Select either 2022-02-22 22:22 or 2011-01-11 11:11 to start the RTC (RealTime Clock) test pattern. Check the system time from either the LCD idle screen or in the web portal System StatusSystem InformationGeneral page. To revert back to the correct time, manually reboot the device.
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Web Info SSH Switch
Hardware Testing Select Test SVIP to verify hardware connections within the device. The result will display on the LCD when the test is complete.
Protocol: Web access protocol (HTTP/ HTTPS). HTTPS is used by default. Port: Web access port number, which is 8089 by default.
Enable SSH Disable SSH SSH access is disabled by default
Use the LED Indicators
The UCM6301/6302 has LED indicators on the network port to display connection status and the following picture shows the other ports status.
Figure 10: Ports Status
The UCM6304/6308 has LED indicators in the front to display connection status. The following table shows the status definitions.
Table 4: UCM6304/6308 LED Indicators
LED Indicator
LED Status
Power 1/Power 2 PoE LAN WAN USB SD FXS ports FXO ports
Solid: Connected Fast Blinking: Data Transferring Slow Blinking: Trying to connect OFF: Not Connected
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Using the Web UI
Accessing the Web UI
The UCM's web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow users to configure the device through a web browser such Microsoft IE (version 8+), Mozilla Firefox, Google Chrome, etc. To access the UCM's web portal, follow the steps below:
Figure 11: UCM6302 Web GUI Login Page
1. Make sure your computer is on the same network as the UCM. 2. Make sure that the UCM's IP address is displayed on its LCD. 3. Enter the UCM's IP address into a web browsers' address bar. The login page should appear (please
see the above image). 4. Enter default administrator username "admin" and password can be found on the sticker at the back of
the UCM. ---------------------------------------------------------------------------------------------------------------------------------------------------
Note: By default, the UCM630X has Redirect From Port 80 enabled. As such, if users type in the UCM630X IP address in the web browser, the web page will be automatically redirected to the page using HTTPS and port 8089. For example, if the LCD shows 192.168.40.167, and 192.168.40.167 is entered into the web browser, the web page will be redirected to: https://192.168.40.167:8089 The option Redirect From Port 80 can be found under the UCM630X Web GUISystem SettingsHTTP Server. ---------------------------------------------------------------------------------------------------------------------------------------------------
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Setup Wizard
After logging into the UCM web portal for the first time, the setup wizard will guide the user through basic configurations such as time zone, network settings, trunks, and routing rules.
Figure 12: UCM630X Setup Wizard
The setup wizard can be closed and reopened at any time. At the end of the wizard, a summary of the pending configuration changes can be reviewed before applying them.
Main Settings
There are 8 main sections in the web portal to manage various features of the UCM.
System Status: Displays the dashboard, system information, current active calls, and network status. Extensions/Trunks: Manages extensions, trunks, and routing rules. Call Features: Manages various features of the UCM such as the IVR and voicemail. PBX Settings: Manages the settings related to PBX functionality such as SIP settings and interface
settings. System Settings: Manages the settings related to the UCM system itself such as network and security
settings. CDR: Contains the call detail records, statistics, and audio recordings of calls processed by the UCM. Value-Added Features: Manages the settings of features unrelated to core PBX functionality such as Zero
Config provisioning and CRM/PMS integrations. Maintenance: Manages settings and logs related to system management and maintenance such as
user management, activity logs, backup settings, upgrade settings and troubleshooting tools.
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Web GUI Languages Currently the UCM630X series Web GUI supports English, Simplified Chinese, Traditional Chinese, Spanish, French, Portuguese, Russian, Italian, Polish, German etc.
Users can select the UCM's web UI display language in the top-right corner of the page.
Figure 13: UCM630X Web GUI Language
Web GUI Search Bar Users can search for options in the web portal with the search bar on the top right of the page.
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Figure 14: Web GUI Search Bar
Saving and Applying Changes
After making changes to a page, click on the "Save" button to save them and then the "Apply Changes" button that finalizes the changes. If a modification requires a reboot, a prompt will appear asking to reboot the device.
Setting Up an Extension
Power on the UCM630X and your SIP endpoint. Connect both devices to the same network and follow the steps below to set up an extension.
1. Log into the UCM web portal and navigate to Extension/TrunkExtensions 2. Click on the "Add" button to start creating a new extension. The Extension and SIP/IAX Password
information will be used to register to this extension. To set up voicemail, the Voicemail Password will be required. 3. To register an endpoint to this extension, go into your endpoint's web UI and edit the desired account. Enter the newly created extension's number, SIP user ID, and password into their corresponding fields on the endpoint. Enter the UCM's IP address into the SIP server field. If setting up voicemail, enter *97 into the Voice Mail Access Number field. This field may be named differently on other devices. 4. To access the extension's voicemail, use the newly registered extension to dial *97 and access the personal voicemail system. Once prompted, enter the voicemail password. If successful, you will now be prompted with various voicemail options. 5. You have now set up an extension on an endpoint.
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SYSTEM SETTINGS
This section will explain the available system-wide parameters and configuration options on the UCM630X series. This includes settings for the following items: General Settings, HTTP server, network Settings, OpenVPN, DDNS Settings, Security Settings, LDAP server, Time settings, Email settings and TR-069.
General Settings
System administrators can prevent the UCM from making calls and/or writing to the data partition (e.g., CDR, recordings, etc.) once the system reaches a specified threshold of storage usage and CPU usage respectively. These options are located in the System Settings General Settings page.
Figure 15: General Settings Interface
Table 5: General Settings Parameters
General Settings
Device Name
Configure the name of the UCM.
Enable CPU Flow Control
Enables the CPU flow control.
Used to set the threshold generated by the CPU Flow Control. When the
CPU Flow Control Threshold system CPU reaches the threshold, it will prohibit the new calls. Default value is 90%.
Data Partition Write Threshold
Used to set a threshold to stop writing data partition. When the disk data partition reaches the threshold configured, the data partition writing will be stopped. Default value is 90%.
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HTTP Server
The UCM630X's embedded web server responds to HTTPS GET/POST requests and allows users to configure the UCM via web browsers such as Microsoft IE, Mozilla Firefox, and Google Chrome. By default, users can access the UCM by just typing its IP address into a browser address bar. The browser will automatically be redirected to HTTPS using port 8089. For example, typing in "192.168.40.50" into the address bar will redirect the browser to "https://192.168.40.50:8089". This behavior can be changed in the System SettingsHTTP Server page.
Table 6: HTTP Server Settings
UCM Web Settings
Redirect From Port 80
Toggles automatic redirection to UCM's web portal from port 80. If disabled, users will need to manually add the UCM's configured HTTPS port to the server address when accessing the UCM web portal via browser. Default is "Enabled".
Port
Specifies the port number used to access the UCM HTTP server. Default is "8089".
Enable IP Address Whitelist
If enabled, only the server addresses in whitelist will be able to access the UCM's web portal. It is highly recommended to add the IP address currently used to access the UCM web page before enabling this option. Default is "Disabled".
Permitted IP(s)
List of addresses that can access the UCM web portal. Ex: 192.168.6.233 / 255.255.255.255
External Host
Configure a URL and port (optional) used to access the UCM web portal or a public link to the video conference room if the UCM is behind NAT.
GS Wave Settings
External Host
Configure a URL and port (optional) used to access the UCM web portal or a public link to the video conference room if the UCM is behind NAT.
Port
The port to access Wave Web and Wave Mobile. If behind NAT, please make sure to map the external port to this port.
Certificate Settings
Certificate Options
Selects the method of acquiring SSL certificates for the UCM web server. Two methods are currently available: Upload Certificate: Upload the appropriate files from one's own PC. Request Certificate: Enter the domain for which to request a
certificate for from "Let's Encrypt".
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TLS Private Key TLS Cert Domain
Uploads the private key for the HTTP server. Note: Key file must be under 2MB in file size and in *.pem format. File name will automatically be changed to "private.pem". Uploads the certificate for the HTTP server. Note: Certificate must be under 2MB in file size and in *.pem format. This will be used for TLS connections and contains private key for the client and signed certificate for the server. Enter the domain to request the certificate for and click on
to request the certificate.
If the protocol or port has been changed, the user will be logged out and redirected to the new URL.
Network Settings
After successfully connecting the UCM630X to the network for the first time, users could login the Web GUI and go to System SettingsNetwork Settings to configure the network parameters for the device.
UCM630X supports Route/Switch/Dual mode functions.
In this section, all the available network setting options are listed for all models. Select each tab in Web GUISystem SettingsNetwork Settings page to configure LAN settings, WAN settings, 802.1X and Port Forwarding.
Basic Settings
Please refer to the following tables for basic network configuration parameters on UCM6301, UCM6302, UCM6304 and UCM6308, respectively.
Method
Table 7: UCM630X Network SettingsBasic Settings
Select "Route", "Switch" or "Dual" mode on the network interface of UCM630X. The default setting is "Route". Route
WAN port will be used for uplink connection. LAN port will function similarly to a regular router port. Switch WAN port will be used for uplink connection. LAN port will be used as a bridge for connections. Dual Both WAN and LAN ports will be used for uplink connections labeled as LAN2 and
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LAN1, respectively. The port selected as the Default Interface will need to have a gateway IP address configured if it is using a static IP.
MTU
Specifies the maximum transmission unit value. Default is 1492.
IPv4 Address
Preferred DNS If configured, this will be used as the Primary DNS server.
Server
WAN (when "Method" is set to "Route")
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
DNS Server 1 Enter the DNS server 1 address for static IP settings.
DNS Server 2 Enter the DNS server 2 address for static IP settings.
Username
Enter the username to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS 802.1Q/VLAN Tag
Assign the VLAN tag of the layer 2 QoS packets for WAN port. The default value is 0.
Layer 2 QoS 802.1p Priority Value
Assign the priority value of the layer 2 QoS packets for WAN port. The default value is 0.
LAN (when Method is set to "Route")
IP Address
Enter the IP address assigned to LAN port. The default setting is 192.168.2.1.
Subnet Mask Enter the subnet mask. The default setting is 255.255.255.0.
DHCP Server Enable or disable DHCP server capability. The default setting is "Yes".
Enable
DNS Server 1 Enter DNS server address 1. The default setting is 8.8.8.8.
DNS Server 2 Enter DNS server address 2. The default setting is 208.67.222.222.
Allow IP Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.
Address From
Allow IP Address To
Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.
Default IP Lease Time
Enter the IP lease time (in seconds). The default setting is 43200.
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LAN (when Method is set to "Switch")
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
DNS Server 1 Enter the DNS server 1 address for static IP settings.
DNS Server 2 Enter the DNS server 2 address for static IP settings.
Username
Enter the username to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS 802.1Q/VLAN Tag
Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0.
Layer 2 QoS 802.1p Priority Value
Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0.
LAN 1 / LAN 2 (when Method is set to "Dual")
Default Interface
If "Dual" is selected as "Method", users will need assign the default interface to be LAN 1 (mapped to UCM6302 WAN port) or LAN 2 (mapped to UCM6302 LAN port) and then configure network settings for LAN 1/LAN 2. The default interface is LAN 2.
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.
Gateway IP
Enter the gateway IP address for static IP settings when the port is assigned as default interface. The default setting is 0.0.0.0.
DNS Server 1 Enter the DNS server 1 address for static IP settings.
DNS Server 2 Enter the DNS server 2 address for static IP settings.
Username
Enter the username to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS 802.1Q/VLAN Tag
Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0.
Layer 2 QoS 802.1p Priority Value
Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0.
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IPv6 Address
WAN (when "Method" is set to "Route")
IP Method
Select Auto or Static. The default setting is Auto
IP Address
Enter the IP address for static IP settings.
IP Prefixlen Enter the Prefix length for static settings. Default is 64
DNS Server 1 Enter the DNS server 1 address for static settings.
DNS Server 2 Enter the DNS server 2 address for static settings.
LAN (when Method is set to "Route")
DHCP Server
Select Disable, Auto or DHCPv6. Disable: the DHCPv6 server is disabled. Auto: Stateless address auto configuration using NDP protocol. DHCPv6: Stateful address auto configuration using DHCPv6 protocol.
The default setting is Disabled.
DHCP Prefix Enter DHCP prefix. (Default is 2001:db8:2:2::)
DHCP prefixlen
Enter the Prefix length for static settings. Default is 64
DNS Server 1 Enter the DNS server 1 address for static settings. Default is (2001:4860:4860::8888 )
DNS Server 2 Enter the DNS server 2 address for static settings. Default is (2001:4860:4860::8844 )
Allow IP Configure starting IP address assigned by the DHCP prefix and DHCP prefixlen.
Address From
Allow IP Address To
Configure the ending IP address assigned by the DHCP Prefix and DHCP prefixlen.
Default IP Lease Time
Configure the lease time (in second) of the IP address.
LAN (when Method is set to "Switch")
IP Method
Select Auto or Static. The default setting is Auto
IP Address
Enter the IP address for static IP settings.
IP Prefixlen Enter the Prefix length for static settings. Default is 64
DNS Server 1 Enter the DNS server 1 address for static settings.
DNS Server 2 Enter the DNS server 2 address for static settings.
LAN 1 / LAN 2 (when Method is set to "Dual")
Default Interface
Users will need assign the default interface to be LAN 1 (mapped to UCM630X WAN port) or LAN 2 (mapped to UCM630X LAN port) and then configure network settings for LAN 1/LAN 2. The default interface is LAN 1.
IP Method
Select Auto or Static. The default setting is Auto
IP Address
Enter the IP address for static IP settings.
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IP Prefixlen DNS Server 1 DNS Server 2
Enter the Prefix length for static settings. Default is 64 Enter the DNS server 1 address for static settings. Enter the DNS server 2 address for static settings.
Method: Route
When the UCM630X has, method set to Route in network settings, WAN port interface is used for uplink connection and LAN port interface is used as a router. Please see a sample diagram below.
Figure 16: UCM6302 Network Interface Method: Route
Method: Switch WAN port interface is used for uplink connection; LAN port interface is used as room for PC connection.
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Figure 17: UCM6302 Network Interface Method: Switch
Method: Dual Both WAN port and LAN port are used for uplink connection. Users will need assign LAN 1 or LAN 2 as the default interface in option "Default Interface" and configure "Gateway IP" if static IP is used for this interface.
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Figure 18: UCM6302 Network Interface Method: Dual
802.1X
IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication mechanism to device before the device can access Internet or other LAN resources. The UCM630X supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure show UCM630X use 802.1X mode "EAP-MD5" on WAN port as client in the network to access Internet.
Figure 19: UCM630X Using 802.1X as Client
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Figure 20: UCM630X Using 802.1X EAP-MD5
The following table shows the configuration parameters for 802.1X on UCM630X. Identity and MD5 password are required for authentication, which should be provided by the network administrator obtained from the RADIUS server. If "EAP-TLS" or "EAP-PEAPv0/MSCHAPv2" is used, users will also need to upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS server.
802.1X Mode
Identity MD5 Password 802.1X CA Certificate 802.1X Client Certificate
Table 8: UCM630X Network Settings802.1X
Select 802.1X mode. The default setting is "Disable". The supported 802.1X mode are: EAP-MD5 EAP-TLS EAP-PEAPv0/MSCHAPv2 Enter 802.1X mode Identity information. Enter 802.1X mode MD5 password information. Select 802.1X certificate from local PC and then upload.
Select 802.1X client certificate from local PC and then upload.
Static Routes
The UCM630X provides users static routing capability that allows the device to use manually configured routes, rather than information only from dynamic routing or gateway configured in the UCM630X Web GUISystem SettingsNetwork SettingsBasic Settings to forward traffic. It can be used to define a route when no other routes are available or necessary, or used in complementary with existing routing on the UCM630X as a failover backup, etc.
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Click on "Add IPv4 Static Route" to create a new IPv4 static route or click on "Add IPv6 Static Route" to create a new IPv6 static route. The configuration parameters are listed in the table below.
Once added, users can select
to edit the static route.
Select
to delete the static route.
Destination
Table 9: UCM630X Network SettingsStatic Routes
Configure the destination IPv4 address or the destination IPv6 subnet for the UCM630X to reach using the static route. Example: IPv4 address - 192.168.66.4 IPv6 subnet - 2001:740:D::1/64
Subnet Mask Gateway Interface
Configure the subnet mask for the above destination address. If left blank, the default value is 255.255.255.255. Example: 255.255.255.0
Configure the IPv4 or IPv6 gateway address so that the UCM630X can reach the destination via this gateway. Gateway address is optional. Example: 192.168.40.5 or 2001:740:D::1
Specify the network interface on the UCM630X to reach the destination using the static route. LAN interface is eth0; WAN interface is eth1.
Static routes configuration can be reset from LCD menuNetwork Menu.
The following diagram shows a sample application of static route usage on UCM6304.
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UCM6304
Figure 21: UCM6304 Static Route Sample
The network topology of the above diagram is as below:
Network 192.168.69.0 has IP phones registered to UCM6304 LAN 1 address Network 192.168.40.0 has IP phones registered to UCM6304 LAN 2 address Network 192.168.66.0 has IP phones registered to UCM6304 via VPN Network 192.168.40.0 has VPN connection established with network 192.168.66.0
In this network, by default the IP phones in network 192.168.69.0 are unable to call IP phones in network 192.168.66.0 when registered on different interfaces on the UCM6304. Therefore, we need configure a static route on the UCM6304 so that the phones in isolated networks can make calls between each other.
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Figure 22: UCM6304 Static Route Configuration
Port Forwarding
The UCM network interface supports router function which provides users the ability to do port forwarding. If LAN mode is set to "Route" under Web GUISystem SettingsNetwork SettingsBasic Settings page, port forwarding is available for configuration.
The port forwarding configuration is under Web GUISystem SettingsNetwork SettingsPort Forwarding page. Please see related settings in the table below.
WAN Port LAN IP
Table 10: UCM630X Network SettingsPort Forwarding
Specify the WAN port number or a range of WAN ports. Unlimited number of ports can be configured.
Note: When it is set to a range, WAN port and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000 will be port forwarding to LAN port 1000. Specify the LAN IP address.
Specify the LAN port number or a range of LAN ports.
LAN Port
Note: When it is set to a range, WAN port and LAN port must be configured with the same range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access
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from WAN port will be forwarded to the LAN port with the same port number, for example, WAN port 1000 will be port forwarding to LAN port 1000.
Select protocol type "UDP Only", "TCP Only" or "TCP/UDP" for the forwarding in Protocol Type
the selected port. The default setting is "UDP Only".
The following figures demonstrate a port forwarding example to provide phone's Web GUI access to public side.
UCM630X network mode is set to "Route". UCM630X WAN port is connected to uplink switch, with a public IP address configured, e.g. 1.1.1.1. UCM630X LAN port provides DHCP pool that connects to multiple phone devices in the LAN network
192.168.2.x. The UCM60X is used as a router, with gateway address 192.168.2.1. There is a GXP2160 connected under the LAN interface network of the UCM630X. It obtains IP address
192.168.2.100 from UCM630X DHCP pool. On the UCM630X Web GUISystem SettingsNetwork SettingsPort Forwarding, configure a port
forwarding entry as the figure shows below. Click on
WAN Port: This is the port opened on the WAN side for access purpose. LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM630X. LAN Port: This is the port opened on the GXP2160 side for access purpose. Protocol Type: We select TCP here for Web GUI access using HTTP.
Figure 23: Create New Port Forwarding
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Figure 24: UCM630X Port Forwarding Configuration
This will allow users to access the GXP2160 Web GUI from public side, by typing in public IP address (example: 1.1.1.1:8088).
Figure 25: GXP2160 Web Access using UCM6302 Port Forwarding
ARP Settings The ARP settings can be configured under Web GUISystem SettingsNetwork SettingsARP Settings
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ARP GC Threshold 1 ARP GC Threshold 2
ARP GC Threshold 3
Table 11: ARP Settings
Minimum number of entries to keep. Garbage collector will not purge entries if there are fewer than this number. The default value is 128.
Threshold when garbage collector becomes more aggressive about purging entries. Entries older than 5 seconds will be cleared when over this number. The default value is 512. Maximum number of non-PERMANENT neighbor entries allowed. Increase this when using large numbers of interfaces and when communicating with large numbers of directly connected peers. The default value is 1024.
OpenVPN®
OpenVPN® settings allow the users to configure UCM630X to use VPN features, the following table gives details about the various options in order to configure the UCM as OpenVPN Client.
OpenVPN® Enable
Configuration Method
OpenVPN® Server Address
Table 12: UCM630X System SettingsNetwork SettingsOpenVPN®
Enable / Disable the OpenVPN® feature.
Select OpenVPN® configuration method. Manual Configuration: Allows to configure OpenVPN® settings manually. Upload Configuration File: Allows to upload. ovpn and .conf files to the UCM and to automatically configure OpenVPN® settings.
Configures the hostname/IP and port of the server. For example: 192.168.1.2:22
OpenVPN® Server Protocol
OpenVPN® Device mode
OpenVPN® Use Compression Enable Weak SSL Ciphers
Specify the protocol user, user should use the same settings as used on the server Use the same setting as used on the server.
Dev TUN: Create a routed IP tunnel. Dev TAP: Create an Ethernet tunnel.
Compress tunnel packets using the LZO algorithm on the VPN link. Do not enable this unless it is also enabled in the server config file.
Either to enable the Weak SSL ciphers or not.
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OpenVPN® Encryption Algorithm OpenVPN® CA Cert OpenVPN® Client Cert OpenVPN® Client Key
Specify the cryptographic cipher. Users should make sure to use the same setting that they are using on the OpenVPN server. Upload as SSL/TLS root certificate. This file will be renamed as `ca.crt' automatically. Upload a client certificate. This file will be renamed as `cliend.crt' automatically.
Upload a client private key. This file will be renamed as `client.key' automatically.
Figure 26: Open VPN® Feature on the UCM630X
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DDNS Settings
DDNS setting allows user to access UCM630X via domain name instead of IP address. The UCM supports DDNS service from the following DDNS provider:
dydns.org noip.com freedns.afraid.org zoneedit.com oray.net
Here is an example of using noip.com for DDNS.
1. Register domain in DDNS service provider. Please note the UCM630X needs to have public IP access.
Figure 27: Register Domain Name on noip.com
2. On Web GUISystem SettingsNetwork SettingsDDNS Settings, enable DDNS service and configure username, password, and host name.
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Figure 28: UCM630X DDNS Setting
3. Now you can use domain name instead of IP address to connect to the UCM630X Web GUI.
Figure 29: Using Domain Name to Connect to UCM630X
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Security Settings
The UCM630X provides users firewall security configurations to prevent certain malicious attack to the UCM630X system. Users could configure to allow, restrict, or reject specific traffic through the device for security and bandwidth purpose. The UCM630X also provides Fail2ban feature for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE. To configure firewall settings in the UCM630X, go to Web GUISystem SettingsSecurity Settings page.
Static Defense
Under Web GUISystem SettingsSecurity SettingsStatic Defense page, users will see the following information:
Current service information with port, process, and type. Typical firewall settings. Custom firewall settings.
The following table shows a sample current service status running on the UCM630X.
Port 7777 389 6060 5060 4569 38563 10000 10001 10002 10003 10004 10005 10006 10007 10010
Table 13: UCM630X FirewallStatic DefenseCurrent Service
Process
Type
Protocol or Service
Asterisk
TCP/IPv4 SIP
Slapd
TCP/IPv4 LDAP
zero_config
UDP/IPv4 UCM630X zero_config service
Asterisk
UDP/IPv4 SIP
Asterisk
UDP/IPv4 SIP
Asterisk
udp/ipv4 SIP
gs_avs
udp/ipv4 gs_avs
gs_avs
udp/ipv4 gs_avs
gs_avs
udp/ipv4 gs_avs
gs_avs
udp/ipv4 gs_avs
gs_avs
udp/ipv4 gs_avs
gs_avs
udp/ipv4 gs_avs
gs_avs
udp/ipv4 gs_avs
gs_avs
udp/ipv4 gs_avs
gs_avs
udp/ipv4 gs_avs
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10012 10013 10014 10015 10018 10019 10020 6066 3306 45678 8439 8088 8888 25 636 4569 42050 7681
gs_avs gs_avs gs_avs gs_avs gs_avs gs_avs gs_avs Python Mysqld Python Lighttpd asterisk Pbxmid Master Slapd asterisk asterisk Pbxmid
udp/ipv4 udp/ipv4 udp/ipv4 udp/ipv4 udp/ipv4 udp/ipv4 udp/ipv4 udp/ipv4 tcp/ipv4 udp/ipv4 tcp/ipv4 tcp/ipv4 tcp/ipv4 tcp/ipv4 tcp/ipv4 udp/ipv6 udp/ipv6 tcp/ipv4
gs_avs gs_avs gs_avs gs_avs gs_avs gs_avs gs_avs python mysqld python HTTP SIP pbxmid master SLDAP SIP SIP pbxmid
For typical firewall settings, users could configure the following options on the UCM630X.
Ping Defense Enable
SYN-Flood Defense Enable
Table 14: Typical Firewall Settings
If enabled, ICMP response will not be allowed for Ping request. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN (UCM630X) interface. Allows the UCM630X to handle excessive amounts of SYN packets from one source and keep the web portal accessible. There are two options available and only one of these options may be enabled at one time. eth(0)LAN defends against attacks directed to the LAN IP address of the
UCM630X. eth(1)WAN defends against attacks directed to the WAN IP address of the
UCM630X.
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Ping-of-Death Defense Enable
SYN Flood Defense will limit the amount of SYN packets accepted by the UCM from one source to 10 packets per second. Any excess packets from that source will be discarded.
Enable to prevent Ping-of-Death attack to the device. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN (UCM630X) interface.
Under "Custom Firewall Settings", users could create new rules to accept, reject or drop certain traffic going through the UCM630X. To create new rule, click on "Create New Rule" button and a new window will pop up for users to specify rule options.
Right next to "Create New Rule" button, there is a checkbox for option "Reject Rules". If it is checked, all the rules will be rejected except the firewall rules listed below. In the firewall rules, only when there is a rule that meets all the following requirements, the option "Reject Rules" will be allowed to check: Action: "Accept" Type "In" Destination port is set to the system login port (e.g., by default 8089) Protocol is not UDP
Figure 30: Create New Firewall Rule
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Table 15: Firewall Rule Settings
Rule Name
Specify the Firewall rule name to identify the firewall rule.
Action Type Interface
Select the action for the Firewall to perform. ACCEPT REJECT DROP
Select the traffic type. IN
If selected, users will need specify the network interface "LAN" or "WAN" (for UCM630X) for the incoming traffic. OUT
Select the interface to use the Firewall rule.
Service
Select the service type. FTP SSH Telnet HTTP LDAP Custom
If "Custom" is selected, users will need specify Source (IP and port), Destination (IP and port) and Protocol (TCP, UDP or Both) for the service. Please note if the source or the destination field is left blank, it will be used as "Anywhere".
Source IP Address and Port
Configure a source subnet and port. If set to "Anywhere" or left empty, traffic from all addresses and ports will be accepted. A single port or a range of ports can be specified (e.g., 10000, 10000-20000).
Destination IP Address and Port
Configure a destination subnet and port. If set to "Anywhere" or left empty, traffic can be sent to all addresses and ports. A single port or a range of ports can be specified (e.g., 10000, 10000-20000).
Protocol
Select the protocol for the rule to be used.
Save the change and click on "Apply" button. Then submit the configuration by clicking on "Apply Changes" on the upper right of the web page. The new rule will be listed at the bottom of the page with sequence number, rule name, action, protocol, type, source, destination, and operation. More operations below:
Click on
to edit the rule.
Click on
to delete the rule.
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Dynamic Defense
Dynamic defense is supported on the UCM630X series. It can blacklist hosts dynamically when the LAN mode is set to "Route" under Web GUISystem SettingsNetwork SettingsBasic Settings page. If enabled, the traffic coming into the UCM630X can be monitored, which helps prevent massive connection attempts or brute force attacks to the device. The blacklist can be created and updated by the UCM630X firewall, which will then be displayed in the web page. Please refer to the following table for dynamic defense options on the UCM630X.
Dynamic Defense Enable Blacklist Update Interval
Connection Threshold
Dynamic Defense Whitelist
Table 16: UCM630X Firewall Dynamic Defense
Enable dynamic defense. The default setting is disabled.
Configure the blacklist update time interval (in seconds). The default setting is 120. Configure the connection threshold. Once the number of connections from the same host reaches the threshold, it will be added into the blacklist. The default setting is 100. Allowed IPs and ports range, multiple IP addresses and port range. For example: 192.168.2.100-192.168.2.105, 1000:9999
The following figure shows a configuration example like this:
If a host at IP address 192.168.5.7 initiates more than 20 TCP connections to the UCM630X it will be added into UCM630X blacklist.
This host 192.168.5.7 will be blocked by the UCM630X for 500 seconds. Since IP range 192.168.5.100-192.168.5.200 is in whitelist, if a host initiates more than 20 TCP connections
to the UCM630X it will not be added into UCM630X blacklist. It can still establish TCP connection with the UCM630X.
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Figure 31: Configure Dynamic Defense
Fail2ban
Fail2Ban feature on the UCM630X provides intrusion detection and prevention for authentication errors in SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within "Max Retry Duration", the UCM630X will act to forbid the host for certain period as defined in "Banned Duration". This feature helps prevent SIP brute force attacks to the PBX system.
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Global Settings Enable Fail2Ban Banned Duration
Figure 32: Fail2ban Settings Table 17: Fail2Ban Settings
Enable Fail2Ban. The default setting is disabled. Please make sure both "Enable Fail2Ban" and "Asterisk Service" are turned on to use Fail2Ban for SIP authentication on the UCM630X. Configure the duration (in seconds) for the detected host to be banned. The default setting is 600. If set to 0, the host will be always banned.
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Max Retry Duration
Within this duration (in seconds), if a host exceeds the max times of retry as defined in "MaxRetry", the host will be banned. The default setting is 600.
MaxRetry
Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The default setting is 5.
Fail2Ban Whitelist
Configure IP address, CIDR mask or DNS host in the whitelist. Fail2Ban will not ban the host with matching address in this list. Up to 20 addresses can be added into the list.
Local Settings
Asterisk Service
Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make sure both "Enable Fail2Ban" and "Asterisk Service" are turned on to use Fail2Ban for SIP authentication on the UCM630X.
Listening Port Number
Configure the listening port number for the service. By default, port 5060 will be used for UDP and TCP, and port 5061 will be used for TCP.
MaxRetry
Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The default setting is 5. Please make sure this option is properly configured as it will override the "MaxRetry" value under "Global Settings".
Enables defense against excessive login attacks to the UCM's web GUI. Login Attack Defense
The default setting is disabled.
Listening Port Number
This is the Web GUI listening port number which is configured under System SettingsHTTP ServerPort. The default is 8089.
MaxRetry
When the number of failed login attempts from an IP address exceeds the MaxRetry number, that IP address will be banned from accessing the Web GUI.
Blacklist
Blacklist
Users will be able to view the IPs that have been blocked by UCM.
SSH Access
SSH switch now is available via Web GUI and LCD. User can enable or disable SSH access directly from Web GUI or LCD screen. For web SSH access, please log in UCM630X web interface and go to Web GUISystem SettingsSecurity SettingsSSH Access.
The "Enable SSH access" option is for system debugging. If you enable this option, the system will allow SSH access. The SSH connection also requires the username and password of the super administrator. This option is turned off by default. It is recommended to turn off this option when debugging is not required.
Enable the "Enable remote SSH (via GDMS)" option, the system will allow remote SSH access via the GDMS platform. This option is turned off by default, and it is strongly recommended to turn off this option when remote troubleshooting is not required.
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Enable SSH Access
Enable Remote SSH via GDMS
Figure 33: SSH Access Table 18: SSH Access
This option is used for system debugging. Once enabled, UCM will allow SSH access. The SSH connection requires super administrator's username and password. The default setting is "No". It is recommended to set it to "No" if there is no need for debugging. If this option is enabled, remote SSH access will be allowed through the GDMS platform. It is strongly recommended to keep this disabled unless remote troubleshooting is necessary.
LDAP Server
The UCM630X has an embedded LDAP/LDAPS server for users to manage corporate phonebook in a centralized manner.
By default, the LDAP server has generated the first phonebook with PBX DN "ou=pbx,dc=pbx,dc=com" based on the UCM630X user extensions already.
Users could add new phonebook with a different Phonebook DN for other external contacts. For example, "ou=people,dc=pbx,dc=com".
All the phonebooks in the UCM630X LDAP server have the same Base DN "dc=pbx,dc=com".
Term Explanation:
cn= Common Name ou= Organization Unit dc= Domain Component These are all parts of the LDAP data Interchange Format, according to RFC 2849, which is how the LDAP tree is filtered.
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If users have the Grandstream phone provisioned by the UCM630X, the LDAP directory will be set up on the phone and can be used right away for users to access all phonebooks. Additionally, users could manually configure the LDAP client settings to manipulate the built-in LDAP server on the UCM630X. If the UCM630X has multiple LDAP phonebooks created, in the LDAP client configuration, users could use "dc=pbx,dc=com" as Base DN to have access to all phonebooks on the UCM630X LDAP server, or use a specific phonebook DN, for example "ou=people,dc=pbx,dc=com", to access to phonebook with Phonebook DN "ou=people,dc=pbx,dc=com " only.
UCM can also act as a LDAP client to download phonebook entries from another LDAP server. To access LDAP server and client settings, go to Web GUISettingsLDAP Server.
LDAP Server Configurations
The following figure shows the default LDAP server configurations on the UCM630X.
Figure 34: LDAP Server Configurations
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The UCM630X LDAP server supports anonymous access (read-only) by default. Therefore, the LDAP client does not have to configure username and password to access the phonebook directory. The "Root DN" and "Root Password" here are for LDAP management and configuration where users will need provide for authentication purpose before modifying the LDAP information.
The default phonebook list in this LDAP server can be viewed and edited by clicking on phonebook under LDAP Phonebook.
for the first
The UCM630X support secure LDAP (LDAPS) where the communication is encrypted and secure.
Figure 35: Default LDAP Phonebook DN
Figure 36: Default LDAP Phonebook Attributes
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LDAP Phonebook Users could use the default phonebook, edit the default phonebook, add new phonebook, import phonebook on the LDAP server as well as export phonebook from the LDAP server. The first phonebook with default phonebook dn "ou=pbx,dc=pbx,dc=com" displayed on the LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. The contacts information will need to be modified via Web GUIExtension/TrunkExtensions first. The default LDAP phonebook will then be updated automatically.
Figure 37: LDAP ServerLDAP Phonebook
Add new phonebook A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under "LDAP Phonebook" section.
Figure 38: Add LDAP Phonebook
Configure the "Phonebook Prefix" first. The "Phonebook DN" will be automatically filled in. For example, if configuring "Phonebook Prefix" as "people", the "Phonebook DN" will be filled with "ou=people,dc=pbx,dc=com". Once added, users can select to edit the phonebook attributes and contact list (see figure below) or select
to delete the phonebook.
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Figure 39: Edit LDAP Phonebook
Import phonebook from your computer to LDAP server Click on "Import Phonebook" and a dialog will prompt as shown in the figure below.
Figure 40: Import Phonebook
The file to be imported must be a CSV, VCF or XML file with UTF-8 encoding. Users can open the file with Notepad and save it with UTF-8 encoding.
Here is how a sample file looks like. Please note "Account Number" and "Phonebook DN" fields are required. Users could export a phonebook file from the UCM630X LDAP phonebook section first and use it as a sample to start with.
Figure 41: Phonebook CSV File Format
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The Phonebook DN field is the same "Phonebook Prefix" entry as when the user clicks on "Add" to create a new phonebook. Therefore, if the user enters "phonebook" in "Phonebook DN" field in the CSV file, the actual phonebook DN "ou=phonebook,dc=pbx,dc=com" will be automatically created by the UCM630X once the CSV file is imported.
In the CSV file, users can specify different phonebook DN fields for different contacts. If the phonebook DN already exists on the UCM630X LDAP Phonebook, the contacts in the CSV file will be added into the existing phonebook. If the phonebook DN does not exist on the UCM630X LDAP Phonebook, a new phonebook with this phonebook DN will be created.
The sample phonebook CSV file in above picture will result in the following LDAP phonebook in the UCM630X.
Figure 42: LDAP Phonebook After Import
As the default LDAP phonebook with DN "ou=pbx,dc=pbx,dc=com" cannot be edited or deleted in LDAP phonebook section, users cannot import contacts with Phonebook DN field "pbx" if existed in the CSV file.
Export phonebook to your computer from UCM630X LDAP server
Select the checkbox for the LDAP phonebook and then click on "Export Selected Phonebook" to export the selected phonebook. The exported phonebook can be used as a record or a sample CSV, VFC or XML file for the users to add more contacts in it and import to the UCM630X again.
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Figure 43: Export Selected LDAP Phonebook
LDAP Client Configurations
The configuration on LDAP client is useful when you use other LDAP servers. Here we provide an example on how to configure the LDAP client on the UCM.
Assuming the remote server base dn is "dc=pbx,dc=com", configure the LDAP client as follows:
LDAP Server : Enter a name for the remote LDAP server Server Address : Enter the IP address or domain name for remote LDAP server. Base DN: dc=pbx,dc=com Username: Enter username if authentication is required Password: Enter password if authentication is required Filter: Enter the filter. Ex: (|(CallerIDName=%)(AccountNumber=%)) Port: Enter the port number. Ex:389 LDAP Name Attributes: Enter the name attributes for remote server LDAP Number Attributes: Enter the number attributes for remote server
The following figure gives a sample configuration for UCM acting as a LDAP client.
Figure 44: LDAP Client Configurations
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To configure Grandstream IP phones as the LDAP clients for UCM, please refer to the following example:
Server Address: The IP address or domain name of the UCM Base DN: dc=pbx,dc=com Username: Please leave this field empty Password: Please leave this field empty LDAP Name Attribute: CallerIDName Email Department FirstName LastName LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax LDAP Number Filter: (AccountNumber=%) LDAP Name Filter: (CallerIDName=%) LDAP Display Name: AccountNumber CallerIDName LDAP Version: If existed, please select LDAP Version 3 Port: 389
The following figure shows the configuration information on a Grandstream GXP2170 to successfully use the LDAP server as configured in [Figure 34: LDAP Server Configurations].
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Figure 45: GXP2170 LDAP Phonebook Configuration
Time Settings
Automatic Date and Time
The current system time on the UCM630X can be found under Web GUISystem StatusDashboardPBX Status. To configure the UCM630X to update time automatically, go to Web GUISystem SettingsTime SettingsAutomatic date and Time. ---------------------------------------------------------------------------------------------------------------------------------------------------
Note: The configurations under Web GUISettingsTime Settings Automatic date and Time page require reboot to take effect. Please consider configuring auto time updating related changes when setting up the UCM630X for the first time to avoid service interrupt after installation and deployment in production. --------------------------------------------------------------------------------------------------------------------------------------------------
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Remote NTP Server Enable DHCP Option 2 Enable DHCP Option 42 Time Zone
Table 19: Time Auto Updating
Specify the URL or IP address of the NTP server for the UCM630X to synchronize the date and time. The default NTP server is pool.ntp.org.
If set to "Yes", the UCM630X can get provisioned for Time Zone from DHCP Option 2 in the local server automatically. The default setting is "Yes".
If set to "Yes", the UCM630X can get provisioned for NTP Server from DHCP Option 42 in the local server automatically. This will override the manually configured NTP Server. The default setting is "Yes".
Select the proper time zone option so the UCM630X can display correct time accordingly.
Set Date and Time
To manually set the time on the UCM630X, go to Web GUISystem SettingsTime SettingsSet Date and Time. The format is YYYY-MM-DD HH:MM:SS.
Figure 46: Set Time Manually
---------------------------------------------------------------------------------------------------------------------------------------------------
Note: Manually setup time will take effect immediately after saving and applying change in the Web GUI. If users would like to reboot the UCM630X and keep the manually setup time setting, please make sure "Remote NTP Server", "Enable DHCP Option 2" and "Enable DHCP Option 42" options under Web GUISettingsTime SettingsAuto Time Updating page are unchecked or set to empty. Otherwise, time auto updating settings in this page will take effect after reboot. --------------------------------------------------------------------------------------------------------------------------------------------------
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NTP Server
The UCM630X can be used as an NTP server for the NTP clients to synchronize their time with. To configure the UCM630X as the NTP server, set "Enable NTP server" to "Yes" under Web GUISystem SettingsTime SettingsNTP Server. On the client side, point the NTP server address to the UCM630X IP address or host name to use the UCM630X as the NTP server.
Office Time
On the UCM630X, the system administrator can define "office time", which can be used to configure time condition for extension call forwarding schedule and inbound rule schedule. To configure office time, go to Web GUISystem SettingsTime SettingsOffice Time. Click on "Add" to create an office time.
Start Time End Time Week
Show Advanced Options
Month Day
Figure 47: Create New Office Time Table 20: Create New Office Time
Configure the start time for office hour. Configure the end time for office hour Select the workdays in one week. Check this option to show advanced options. Once selected, please specify "Month" and "Day" below. Select the months for office time. Select the workdays in one month.
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Select "Start Time", "End Time" and the day for the "Week" for the office time. The system administrator can also define month and day of the month as advanced options. Once done, click on "Save" and then "Apply Change" for the office time to take effect. The office time will be listed in the web page as the figure shows below.
Click on
Figure 48: SettingsTime SettingsOffice Time
to edit the office time.
Click on to delete the office time.
Click on "Delete" to delete multiple selected office times at once.
Holiday
On the UCM630X, the system administrator can define "holiday", which can be used to configure time condition for extension call forwarding schedule and inbound rule schedule. To configure holiday, go to Web GUISystem SettingsTime SettingsHoliday. Click on "Add" to create holiday time.
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Figure 49: Create New Holiday
Name Holiday Memo Month Day
Show Advanced Options
Week
Table 21: Create New Holiday
Specify the holiday name to identify this holiday. Create a note for the holiday. Select the month for the holiday. Select the day for the holiday. Check this option to show advanced options. If selected, please specify the days as holiday in one week below. Select the days as holiday in one week.
Enter holiday "Name" and "Holiday Memo" for the new holiday. Then select "Month" and "Day". The system administrator can also define days in one week as advanced options. Once done, click on "Save" and then "Apply Change" for the holiday to take effect. The holiday will be listed in the web page as the figure shows below.
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Click on
Figure 50: SettingsTime SettingsHoliday
to edit the holiday.
Click on
to delete the holiday.
Click on "Delete" to delete multiple selected holidays at once. --------------------------------------------------------------------------------------------------------------------------------------------
Note:
For more details on how to use office time and holiday, please refer to the link below: http://www.grandstream.com/sites/default/files/Resources/office_time_and_holiday_on_ucm6xxx.pdf --------------------------------------------------------------------------------------------------------------------------------------------
Email Settings
Email settings
The Email application on the UCM630X can be used to send out alert event Emails, Voicemail (Voicemail-ToEmail) etc. The configuration parameters can be accessed via Web GUISystem SettingsEmail SettingsEmail Settings.
Table 22: Email Settings
TLS Enable
Enable or disable TLS during transferring/submitting your Email to another SMTP server. The default setting is "Yes".
Type
Select Email type. MTA: Mail Transfer Agent. The Email will be sent from the
configured domain. When MTA is selected, there is no need to set up SMTP server for it or no user login is required. However, the Emails sent from MTA might be considered as spam by the target SMTP server. Client: Submit Emails to the SMTP server. A SMTP server is required, and users need login with correct credentials.
Domain
Specify the domain name to be used in the Email when using type "MTA".
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SMTP Server
Enable SASL Authentication
Username Password POP/POP3 Server Address POP/POP3 Server Port Display Name Sender
Specify the SMTP server when using type "Client". Enable SASL Authentication. When disabled, UCM will not try to use the username and password for mail client login authentication. Most of the mail server requires login authentication while some others private mail servers allow anonymous login which requires disabling this option to send Email as normal. For Exchange Server, please disable this option. Username is required when using type "Client". Normally it is the Email address. Password to login for the above Username (Email address) is required when using type "Client". Configure the POP/POP3 server address for the configured username Example: pop.gmail.com Configure the POP/POP3 server port for the configured username Example: 995 Specify the display name in the FROM header in the Email. Specify the sender's Email address. For example: pbx@example.mycompany.com.
The following figure shows a sample Email setting on the UCM630X, assuming the Email is using 192.168.6.202 as the SMTP server.
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Figure 51: UCM630X Email Settings
Once the configuration is finished, click on "Test". In the prompt, fill in a valid Email address to send a test Email to verify the Email settings on the UCM630X. Email Templates The Email templates on the UCM630X can be used for email notification, the configuration parameters can be accessed via Web GUISettingsEmail SettingsEmail Templates.
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Figure 52: Email Template
Email Send Log
Under UCM Web GUISystem SettingsEmail SettingsEmail Send Log, users could search, filter and check whether the Email is sent out successfully or not. This page will also display the corresponding error message if the Email is not sent out successfully.
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Figure 53: Email Send Log
Field Start Time End Time Receivers
Send Result Return Code Email Send Module
Table 23: Email Log Display Filter
Description Enter the start time for filter Enter the end time for filter
Enter the email recipient, while searching for multiple recipients, please separate then with comma and no spaces.
Enter the status of the send result to filter with
Enter the email code to filter with Select the email module to filter with from the drop-down list, which contains: All Modules Extension Voicemail Conference Schedule User Password Alert Events CDR Test
Email logs will be shown on bottom of the "Email Send Log" page, as shown on the following figure.
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Figure 54: Email Logs
Below are the codes returned when sending emails and their description:
Code 250 501
535 550
552 553 554
Table 24: Email Codes
Description Mail sent successfully Address format parsing error, 501 will be returned when there are unacceptable characters in the recipient's email address in MTA mode. Please check if the recipient's email address format is correct. The "sender" configured on the client is your mail account. The user name and password verification in the client mode is incorrect. Please check whether the user name and password are configured correctly. Possible reasons: 1. The recipient's mailbox user name does not exist or is in a banned state, please check whether the email recipient is the correct email address. 2. The number of destination addresses sent by the sender exceeds the maximum limit per day and is temporarily blacklisted. Please reduce the sending frequency or try again the next day. 3. The sender's IP does not pass the SPF permission test of the sending domain. Emails sent in MTA mode may return this error code even if they are sent. The sent email is too large or the email attachment type is prohibited The sender and the email account are inconsistent, please configure the sender as your email account correctly. The email was identified as spam. Please reduce the sending frequency or try again the next day
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none
Indicates that there is no return code. If the sending result is "deferred", the general reason is that the mail service area is configured incorrectly. Please check whether the server configuration is correct. If the sending result is "bounced", the general reason is that the receiving email address domain name is wrong, please check whether the email recipient is the correct email address. If it is in MTA mode, please check whether the "domain" is configured to be in the same domain name as the "recipient".
TR-069
To configure TR-069 on Grandstream devices, set following parameters:
Parameter Enable TR-069 ACS URL
Description Toggle it on to enable TR-069. It is enabled by default URL for TR-069 Auto Configuration Servers (ACS), e.g., http://myacs.grandstream.com
TR-069 Username
ACS username for TR-069, must be the same as in the ACS configuration.
TR-069 Password
ACS password for TR-069, must be the same as in the ACS configuration.
Periodic Inform Enable Enables periodic inform. If set to 'Yes', device will send inform packets to the ACS.
Periodic Inform Interval
Periodic time when UCM630X will send inform packets to TR-069 ACS server. This option is specified in seconds.
ACS Connection Request Username
The username for the ACS to connect to UCM.
ACS Connection Request Password
The password for the ACS to connect to UCM.
Connection Request Port
Port for incoming connection requests. The default value is 7547.
CPE Cert File
The Cert file for UCM to connect to the ACS via SSL.
CPE Cert Key
The Cert key for UCM to connect to the ACS via SSL.
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PROVISIONING
Overview
Grandstream SIP Devices can be configured via Web interface as well as via configuration file through TFTP/HTTP/HTTPS download. All Grandstream SIP devices support a proprietary binary format configuration file and XML format configuration file. The UCM630X provides a Plug and Play mechanism to auto-provision the Grandstream SIP devices in a zero-configuration manner by generating XML config file and having the phone to download it within LAN area. This allows users to finish the installation with ease and start using the SIP devices in a managed way.
To provision a phone, three steps are involved, i.e., discovery, configuration, and provisioning. This section explains how Zero Config works on the UCM630X. The settings for this feature can be accessed via Web GUIValue-added FeaturesZero Config.
Configuration Architecture for End Point Device
Started from firmware version 1.0.7.10, the end point device configuration in zero config is divided into the following three layers with priority from the lowest to the highest:
Global This is the lowest layer. Users can configure the most basic options that could apply to all Grandstream SIP devices during provisioning via Zero config.
Model In this layer, users can define model-specific options for the configuration template.
Device This is the highest layer. Users can configure device-specific options for the configuration for individual device here.
Each layer also has its own structure in different levels. Please see figure below. The details for each layer are explained in sections [Global Configuration], [Model configuration] and [Device Configuration].
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Figure 55: Zero Config Configuration Architecture for End Point Device
The configuration options in model layer and device layer have all the option in global layers already, i.e., the options in global layer is a subset of the options in model layer and device layer. If an option is set in all three layers with different values, the highest layer value will override the value in lower layer. For example, if the user selects English for Language setting in Global Policy and Spanish for Language setting in Default Model Template, the language setting on the device to be provisioned will use Spanish as model layer has higher priority than global layer. To sum up, configurations in higher layer will always override the configurations for the same options/fields in the lower layer when presented at the same time.
After understanding the zero-config configuration architecture, users could configure the available options for end point devices to be provisioned by the UCM630X by going through the three layers. This configuration architecture allows users to set up and manage the Grandstream end point devices in the same LAN area in a centralized way.
Auto Provisioning Settings
By default, the Zero Config feature is enabled on the UCM630X for auto provisioning. Three methods of auto provisioning are used.
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Figure 56: UCM630X Zero Config
SIP SUBSCRIBE When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The UCM630X discovers it and then sends a NOTIFY with the XML config file URL in the message body. The phone will then use the path to download the config file generated in the UCM630X and take the new configuration.
DHCP OPTION 66 Route mode needs to be set to use this feature. When the phone restarts (by default DHCP Option 66 is turned on), it will send out a DHCP DISCOVER request. The UCM630X receives it and returns DHCP OFFER with the config server path URL in Option 66, for example, https://192.168.2.1:8089/zccgi/. The phone will then use the path to download the config file generated in the UCM630X.
mDNS When the phone boots up, it sends out mDNS query to get the TFTP server address. The UCM630X will respond with its own address. The phone will then send TFTP request to download the XML config file from the UCM630X.
To start the auto provisioning process, under Web GUIValue-added FeaturesZero ConfigZero Config Settings, fill in the auto provision information.
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Enable Zero Config
Enable Automatic Configuration Assignment
Auto Assign Extension
Figure 57: Auto Provision Settings
Table 25: Auto Provision Settings
Enable or disable the zero-config feature on the PBX. The default setting is enabled. By default, this is disabled. If disabled, when SIP device boots up, the UCM630X will not send the SIP device the URL to download the config file and therefore the SIP device will not be automatically provisioned by the UCM630X.
Note: When disabled, SIP devices can still be provisioned by manually sending NOTIFY from the UCM630X which will include the XML config file URL for the SIP device to download. If enabled, when the device is discovered, the PBX will automatically assign an extension within the range defined in "Zero Config Extension Segment" to the device. The default setting is disabled.
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Zero Config Extension Segment Enable Pick Extension
Pick Extension Segment Pick Extension Period (hour)
Subnet Whitelist
Click on the link "Zero Config Extension Segment" to specify the extension range to be assigned if "Automatically Assign Extension" is enabled. The default range is 5000-6299. Zero Config Extension Segment range can be defined in Web GUIPBX SettingsGeneral SettingsGeneral pageExtension Preference section: "Auto Provision Extensions".
If enabled, the extension list will be sent out to the device after receiving the device's request. This feature is for the GXP series phones that support selecting extension to be provisioned via phone's LCD. The default setting is disabled.
Click on the link "Pick Extension Segment" to specify the extension list to be sent to the device. The default range is 4000 to 4999. Pick Extension Segment range can be defined in Web GUI PBX Settings General Settings General page Extension Preference section: "Pick Extensions".
Specify the number of minutes to allow the phones being provisioned to pick extensions.
This feature allows the UCM to provision devices in different subnets other than UCM network. Enter subnets IP addresses to allow devices within these subnets to be provisioned. The syntax is <IP>/<CIDR>. Examples: 10.0.0.1/8 192.168.6.0/24 Note: Only private IP ranges (10.0.0.0 | 172.16.0.0 | 192.168.0.0) are supported.
Please make sure an extension is manually assigned to the phone or "Automatically Assign Extension" is enabled during provisioning. After the configuration on the UCM630X Web GUI, click on "Save" and "Apply Changes". Once the phone boots up and picks up the config file from the UCM630X, it will take the configuration right away.
Discovery
Grandstream endpoints are automatically discovered after bootup. Users could also manually discover device by specifying the IP address or scanning the entire LAN network. Three methods are supported to scan the devices.
PING ARP
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SIP Message (NOTIFY)
Click on "Auto Discover" under Web GUIValue-added FeaturesZero ConfigZero Config, fill in the "Scan Method" and "Scan IP". The IP address segment will be automatically filled in based on the network mask detected on the UCM630X. If users need scan the entire network segment, enter 255 (for example, 192.168.40.255) instead of a specific IP address. Then click on "Save" to start discovering the devices within the same network. To successfully discover the devices, "Zero Config" needs to be enabled on the UCM630X Web GUIValue-added FeaturesZero ConfigAuto Provisioning Settings.
Figure 58: Auto Discover
The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if assigned), Version, Vendor, Model, Connection Status, Create Config, Options (Edit /Delete /Update /Reboot /Access Device WebGUI) are displayed in the list.
Figure 59: Discovered Devices
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Uploading Devices List
Besides the built-in discovery method on the UCM, users could prepare a list of devices on .CSV file and upload it by clicking on the button "Import", after which a success message prompt should be displayed. Users need to make sure that the CSV file respects the format as shown on the following figure and that the entered information is correct (valid IP address, valid MAC address, device model and an existing account), otherwise the UCM will reject the file and the operation will fail:
Figure 60: Device List - CSV file Sample
Managing Discovered Devices
Sorting: Press or to sort per MAC Address, IP Address, Version, Vendor, Model or Create Config columns from lower to higher or higher to lower, respectively.
Filter: Select a filter
to display corresponding results.
All: Display all discovered devices.
Scan Results: Display only manually discovered devices. [Discovery]
IP Address: Enter device IP and press Search button.
MAC Address: Enter device MAC and press Search button.
Model: Enter a model name and press Search button. Example: GXP2130.
Extension: Enter the extension number and press Search button.
Figure 61: Managing Discovered Devices
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From the main menu of zero config, users can perform the following operations:
Click on
in order to access to the discovery menu as shown on [Discovery] section.
Click on
to add a new device to zero config database using its MAC address.
Click on
to delete selected devices from the zero-config database.
Click on
to modify selected devices.
Click on
to batch update a list of devices, the UCM on this case will send SIP NOTIFY
message to all selected devices in order to update them at once.
Click on
to reboot selected devices (the selected devices, should have been provisioned with
extensions since the phone will authenticate the server which is trying to send it reboot command).
Click on
to clear all devices configurations.
Click on
to upload CSV file containing list of devices.
Click on
to export CSV file containing list of devices. This file can be imported to another UCM
to quickly set it up with the original UCM's devices.
Click on
to copy configuration from one device to another. This can be useful for easily replace
devices and note that this feature works only between devices of same model.
All these operations will be detailed on the next sections.
Global Configuration
Global Policy
Global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN with the UCM630X no matter what the Grandstream device model it is. It is divided into two levels:
Global Policy Global Templates
Note: Global Templates configuration has higher priority to Global Policy configuration.
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Global Policy can be accessed in Web GUIValue-added FeaturesZero ConfigGlobal Policy page. On the top of the configuration table, users can select category in the "Options" dropdown list to quickly navigate to the category. The categories are:
Localization: configure display language, data, and time. Phone Settings: configure dial plan, call features, NAT, call progress tones and etc. Contact List: configure LDAP and XML phonebook download. Maintenance: configure upgrading, web access, Telnet/SSH access and syslog. Network Settings: configure IP address, QoS and STUN settings. Customization: customize LCD screen wallpaper for the supported models. Communication Settings: configure Email and FTP settings
Select the checkbox on the left of the parameter you would like to configure to activate the dropdown list for this parameter.
Figure 62: Global Policy Categories
The following tables list the Global Policy configuration parameters for the SIP end device.
Language settings Language Date and Time Date Format
Table 26: Global Policy Parameters Localization
Select the LCD display language on the SIP end device. Configure the date display format on the SIP end device's LCD.
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Time Format
Configure the time display in 12-hour or 24-hour format on the SIP end device's LCD.
Enable NTP
To enable the NTP service.
NTP Server
Configure the URL or IP address of the NTP server. The SIP end device may obtain the date and time from the server.
NTP Update Interval
Configure the NTP update interval.
Time Zone
Configure the time zone used on the SIP end device.
Enable Daylight Saving Time Select either to enable or disable the DST.
Table 27: Global Policy Parameters Phone Settings
Default Call Settings
Dial Plan
Configure the default dial plan rule. For syntax and examples, please refer to user manual of the SIP devices to be provisioned for more details.
Enable Call Features
When enabled, "Do Not Disturb", "Call Forward" and other call features can be used via the local feature code on the phone. Otherwise, the ITSP feature code will be used.
Use # as Dial Key
If set to "Yes", pressing the number key "#" will immediately dial out the input digits.
Auto Answer by Call-info
If set to "Yes", the phone will automatically turn on the speaker phone to answer incoming calls after a short reminding beep, based on the SIP Call-Info header sent from the server/proxy. The default setting is enabled.
NAT Traversal
Configure if NAT traversal mechanism is activated.
User Random Port
If set to "Yes", this parameter will force random generation of both the local SIP and RTP ports.
General Settings
Call Progress Tones
Configure call progress tones including ring tone, dial tone, second dial tone, message waiting tone, ring back tone, call waiting tone, busy tone and reorder tone using the following syntax: f1=val, f2=val[, c=on1/ off1[- on2/ off2[- on3/ off3]]]; Frequencies are in Hz and cadence on and off are in 10ms). "on" is the period (in ms) of ringing while "off" is the period of silence.
Up to three cadences are supported. Please refer to user manual of the SIP devices to be provisioned for
more details
HEADSET Key Mode
Select "Default Mode" or "Toggle Headset/Speaker" for the Headset key. Please refer to user manual of the SIP devices to be provisioned for more details.
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LDAP Phonebook Source Address Port Base DN Username Password Number Filter Name Filter Version
Name Attribute
Number Attribute
Display Name
Table 28: Global Policy Parameters Contact List
Select "Manual" or "PBX" as the LDAP configuration source. If "Manual" is selected, the LDAP configuration below will be applied to
the SIP end device. If "PBX" is selected, the LDAP configuration built-in from UCM630X
Web GUISystem SettingsLDAP Server will be applied. Configure the IP address or DNS name of the LDAP server. Configure the LDAP server port. The default value is 389. This is the location in the directory where the search is requested to begin. Example: dc=grandstream, dc=com ou=Boston, dc=grandstream, dc=com Configure the bind "Username" for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds. Configure the bind "Password" for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds. Configure the filter used for number lookups. Please refer to user manual for more details. Configure the filter used for name lookups. Please refer to user manual for more details. Select the protocol version for the phone to send the bind requests. The default value is 3. Specify the "name" attributes of each record which are returned in the LDAP search result. Example: gn cn sn description Specify the "number" attributes of each record which are returned in the LDAP search result. Example: telephoneNumber telephoneNumber Mobile Configure the entry information to be shown on phone's LCD. Up to 3 fields can be displayed. Example: %cn %sn %telephoneNumber
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Max Hits Search Timeout Sort Results Incoming Calls Outgoing Calls Lookup Display Name XML Phonebook
Phonebook XML Server
Phonebook Download Interval Remove manually edited entries on download
Specify the maximum number of results to be returned by the LDAP server. Valid range is 1 to 3000. The default value is 50. Specify the interval (in seconds) for the server to process the request and client waits for server to return. Valid range is 0 to 180. Default value is 30. Specify whether the searching result is sorted or not. Default setting is No. Configure to enable LDAP number searching when receiving calls. The default setting is No. Configure to enable LDAP number searching when making calls. The default setting is No. Configures the display name when LDAP looks up the name for incoming call or outgoing call. It must be a subset of the LDAP Name Attributes.
Select the source of the phonebook XML server.
Disable Disable phonebook XML downloading.
Manual Once selected, users need specify downloading protocol HTTP, HTTPS or TFTP and the server path to download the phonebook XML file. The server path could be IP address or URL, with up to 256 characters.
Local UCM Server Once selected, click on the Server Path field to upload the phonebook XML file. Please note after uploading the phonebook XML file to the server, the original file name will be used as the directory name and the file will be renamed as phonebook.xml under that directory.
Configure the phonebook download interval (in Minute). If set to 0, automatic download will be disabled. Valid range is 5 to 720. If set to "Yes", when XML phonebook is downloaded, the entries added manually will be automatically removed.
Upgrade and Provision Firmware Source
Table 29: Global Policy Parameters Maintenance
Firmware source via ZeroConfig provisioning could a URL for external server address, local UCM directory or USB media if plugged in to the UCM630X. Select a source to get the firmware file:
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Upgrade via Server Path File Prefix Config Server Path Allow DHCP Option 43/66
Automatic Upgrade
URL If select to use URL to upgrade, complete the configuration for the following four parameters: "Upgrade Via", "Server Path", "File Prefix" and "File Postfix".
Local UCM Server Firmware can be uploaded to the UCM630X internal storage for firmware upgrade. If selected, click on "Manage Storage" icon next to "Directory" option, upload firmware file and select directory for the end device to retrieve the firmware file.
Local USB Media If selected, the USB storage device needs to be plugged into the UCM630X and the firmware file must be put under a folder named "ZC_firmware" in the USB storage root directory.
Local SD Card Media If selected, an SD card needs to be plugged into the UCM630X and the firmware file must be put under a folder named "ZC_firmware" in the USB storage root directory.
When URL is selected as firmware source, configure upgrade via TFTP, HTTP or HTTPS. When URL is selected as firmware source, configure the firmware upgrading server path. Configure the Config Server Path. When URL is selected as firmware source, configure the firmware file postfix. If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone. If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can be redirected. If enabled, the end point device will automatically upgrade if a new firmware is detected. Users can select automatic upgrading by day, by week or by minute. By week
Once selected, specify the day of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes. By day Once selected, specify the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes.
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Firmware Upgrade Rule Zero Config Web Access Admin Password End-User Password Web Access Mode Web Server Port RTSP Port Enable UPnP Discovery Login Settings User Login Timeout Maximum Consecutive Failed Login Attempts Login Error Lock Time Security
Disable Telnet/SSH
Syslog Syslog Server Syslog Level Send SIP Log
Basic Settings
IP Address
By minute Once selected, specify the interval X that the SIP end device will request for new firmware every X minutes.
Specify how firmware upgrading and provisioning request to be sent. Select either to enable or disable zero config.
Configure the administrator password for admin level login. Configure the end-user password for the end user level login. Select HTTP or HTTPS as the web access protocol. Configure the port for web access. The valid range is 1 to 65535. Configure the RTSP Port. Select either to enable or disable Enable UPnP Discovery Configure the login settings. Configure User Login Timeout.
Configure Maximum Consecutive Failed Login Attempts.
Configure Login Error Lock Time.
Enable Telnet/SSH access for the SIP end device. If the SIP end device supports Telnet access, this option controls the Telnet access of the device; if the SIP end device supports SSH access, this option controls the SSH access of the device.
Configure the URL/IP address for the syslog server. Select the level of logging for syslog. Configure whether the SIP log will be included in the syslog message.
Table 30: Global Policy Parameters Network Settings
Configure how the SIP end device shall obtain the IP address. DHCP or PPPoE can be selected. DHCP
Once selected, users can specify the Host Name (option 12) of the SIP end device as DHCP client, and Vendor Class ID (option 60) used by
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Host Name Vendor Class ID Account ID Password Service Name Advanced Setting Layer 3 QoS
Layer 3 QoS For RTP
Layer 3 QoS For SIP
Layer 2 QoS Tag
Layer 2 QoS Priority Value
STUN Server Keep Alive
Keep Alive Interval
Register Expiration Local SIP Port Local RTP Port Auto On-Hook Timer(s) Ring Timeout SIP Transport Direct IP Call SIP Proxy Compatibility Mode
the client and server to exchange vendor class ID information.
PPPoE Once selected, users need specify the Account ID, Password and Service Name for PPPoE.
Specifies the name of the client. This field is optional but may be required by Internet Service Providers. Used by clients and servers to exchange vendor class ID. Enter the PPPoE account ID. Enter the PPPoE Password. Enter the PPPoE Service Name.
Define the Layer 3 QoS parameter. This value is used for IP Precedence, Diff-Serv or MPLS. Valid range is 0-63. Assign the priority value of the Layer 3 QoS for RTP packets. Valid range is 0 -63. Assign the priority value of the Layer 3 QoS for SIP packets. Valid range is 0 -63. Assign the VLAN Tag of the Layer 2 QoS packets. Valid range is 0 -4095. Assign the priority value of the Layer 2 QoS packets. Valid range is 0-7. Configure the IP address or Domain name of the STUN server. Only nonsymmetric NAT routers work with STUN. Select either to enable or disable Keep Alive. Specify how often the phone will send a blank UDP packet to the SIP server in order to keep the "ping hole" on the NAT router to open. Valid range is 10-160. Specify the Register Expiration. Configure Local SIP Port. Configure Local RTP Port. Configure Auto On-Hook Timer(s). Configure Ring Timeout. Select either UDP, TCP or TLS/TCP as SIP transport protocol. Select either to disable or enable Direct IP Call support.
Select either to disable or enable SIP Proxy Compatibility Mode.
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Unregister On Reboot Whitelist Whitelist SIP Phone Number Whitelist
Select either to disable or enable Unregister On Reboot.
Select either to enable or disable Whitelist Configure the SIP Phone Number Whitelist.
Wallpaper
Table 31: Global Policy Parameters Customization
Check this option if the SIP end device shall use 1024 x 600 resolution for the LCD screen wallpaper.
Screen Resolution 1024 x 600
Source Configure the location where wallpapers are stored.
File If "URL" is selected as source, specify the URL of the wallpaper file. If "Local UCM Server" is selected as source, click to upload wallpaper file to the UCM630X.
Check this option if the SIP end device shall use 800 x 400 resolution for the LCD screen wallpaper.
Screen Resolution 800 x 400
Source Configure the location where wallpapers are stored.
File If "URL" is selected as source, specify the URL of the wallpaper file. If "Local UCM Server" is selected as source, click to upload wallpaper file to the UCM630X.
Check this option if the SIP end device shall use 480 x 272 resolution for the LCD screen wallpaper.
Screen Resolution 480 x 272 Screen Resolution 320 x 240
Source Configure the location where wallpapers are stored.
File If "URL" is selected as source, specify the URL of the wallpaper file. If "Local UCM Server" is selected as source, click to upload wallpaper file to the UCM630X.
Check this option if the SIP end device supports 320 x 240 resolution for the LCD screen wallpaper.
Source Configure the location where wallpapers are stored.
File
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Email Settings
SMTP Settings
FTP FTP
If "URL" is selected as source, specify the URL of the wallpaper file. If "Local UCM Server" is selected as source, click to upload wallpaper file to the UCM630X.
Table 32: Global Policy Parameters Communication Settings
Check this option to configure the email settings that will be sent to the provisioned phones: Server
IP address of the SMTP server Port
SMTP server port From E-Mail address
Email address Sender Username
Username of the sender Password Recovery Email
Email where recovered password will be sent Alarm receive Email 1
Email address where alarms notifications will be sent Alarm receive Email 1
Email address where alarms notifications will be sent Enable SSL
Enable SSL protocol for SMTP
Check this option to configure the FTP settings that will be sent to the provisioned phones: Storage Server Type
Either FTP or Central Storage Server
FTP server address
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Port FTP port to be used
Username FTP username
Path FTP Directory path
Global Templates
Global Templates can be accessed in Web GUIValue-added FeaturesZero ConfigGlobal Templates. Users can create multiple global templates with different sets of configurations and save the templates. Later on, when the user configures the device in Edit Device dialogAdvanced Settings, the user can select to use one of the global templates for the device. Please refer to section [Manage Devices] for more details on using the global templates.
When creating global template, users can select the categories and the parameters under each category to be used in the template. The global policy and the selected global template will both take effect when generating the config file. However, the selected global template has higher priority to the global policy when it comes to the same setting option/field. If the same option/field has different value configured in the global policy and the selected global template, the value for this option/field in the selected global template will override the value in global policy.
Click on "Add" to add a global template. Users will see the following configurations.
Template Name Description Active
Table 33: Create New Template
Create a name to identify this global template. Provide a description for the global template. This is optional. Check this option to enable the global template.
Click on
to edit the global template.
The window for editing global template is shown in the following figure. In the "Options" field, after entering the option name key word, the options containing the key word will be listed. Users could then select the options to be modified under the global template.
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Figure 63: Edit Global Template
The added options will show in the list. Users can then enter or select value for each option to be used in the global template. On the left side of each added option, users can click on to delete this option from the template. On the right side of each option, users can click on to reset the option value to the default value.
Click on "Save" to save this global template. The created global templates will show in the Web GUIValue-added FeaturesZero ConfigGlobal
Templates page. Users can click on at once.
to delete the global template or delete multiple selected templates
Click on "Toggle Selected Template(s)" to toggle the status between enabled/disabled for the selected templates.
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Model configuration
Model templates
Model layer configuration allows users to apply model-specific configurations to different devices. Users could create/edit/delete a model template by accessing Web GUI, page Value-added FeaturesZero ConfigModel Templates. If multiple model templates are created and enabled, when the user configures the device in Edit Device dialogAdvanced Settings, the user can select to use one of the model templates for the device. Please refer to section [Manage Devices] for more details on using the model template.
For each created model template, users can assign it as default model template. If assigned as default model template, the values in this model template will be applied to all the devices of this model. There is always only one default model template that can be assigned at one time on the UCM630X.
The selected model template and the default model template will both take effect when generating the config file for the device. However, the model template has higher priority to default model template when it comes to the same setting option/field. If the same option/field has different value configured in the default model template and the selected model template, the value for this option/field in the selected model template will override the value in default model template.
Click on "Add" to add a model template.
Model
Table 34: Create New Model Template
Select a model to apply this template. The supported Grandstream models are listed in the dropdown list for selection.
Template Name
Create a name for the model template.
Description Default Model Template Active
Enter a description for the model template. This is optional. Select to assign this model template as the default model template. The value of the option in default model template will be overridden if other selected model template has a different value for the same option. Check this option to enable the model template.
Click on
to edit the model template.
The editing window for model template is shown in the following figure. In the "Options" field, enter the option name key word, the option that contains the key word will be listed. User could then select the option to be modified under the model template.
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Once added, the option will be shown in the list below. On the left side of each option, users can click on to remove this option from the model template. On the right side of each option, users can click on to reset the option to the default value.
User could also click on "Add New Field" to add a P value number and the value to the configuration. The following figure shows setting P value "P1362" to "en", which means the display language on the LCD is set to English. For P value information of different models, please refer to configuration template here http://www.grandstream.com/support/tools.
Figure 64: Edit Model Template
Click on Save when done. The model template will be displayed on Web GUIValue-added FeaturesZero ConfigModel Templates page.
Click on to delete the model template or click on "Delete Selected Templates" to delete multiple selected templates at once.
Click on "Toggle Selected Template(s)" to toggle the status between enabled/disabled for the selected model templates.
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Model Update
UCM630X zero config feature supports provisioning all models of Grandstream SIP end devices including OEM device models.
OEM Models
Users can associate OEM device models with their original Grandstream-branded models, allowing these OEM devices to be provisioned appropriately.
Click on
button.
In the Source Model field, select the Grandstream device that the OEM model is based on from the dropdown
list.
For the Destination Model and Destination Vendor field, enter the custom OEM model name and vendor name.
The newly added OEM model should now be selectable as an option in Model fields.
Figure 65: OEM Models
Model Template Package List
Templates for most of the Grandstream models are built in with the UCM630X already. Templates for GS Wave and Grandstream surveillance products require users to download and install under Web GUIValue-added FeaturesZero ConfigModel Update first before they are available in the UCM630X for selection. After downloading and installing the model template to the UCM630X, it will show in the dropdown list for "Model" selection when editing the model template.
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Click on to download the template.
Click on to upgrade the model template. Users will see this icon available if the device model has template updated in the UCM630X.
Figure 66: Template Management
Upload Model Template Package
In case the UCM630X is placed in the private network and Internet access is restricted, users will not be able to get packages by downloading and installing from the remote server. Model template package can be manually uploaded from local device through Web GUI. Please contact Grandstream customer support if the model package is needed for manual uploading.
Figure 67: Upload Model Template Manually
Device Configuration
On Web GUI, page Value-added FeaturesZero ConfigZero Config, users could create new device, delete existing device(s), make special configuration for a single device, or send NOTIFY to existing device(s).
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Create New Device
Besides configuring the device after the device is discovered, users could also directly create a new device and configure basic settings before the device is discovered by the UCM630X. Once the device is plugged in, it can then be discovered and provisioned. This gives the system administrator adequate time to set up each device beforehand.
Click on "Add" and the following dialog will show. Follow the steps below to create the configurations for the new device.
1. Firstly, select a model for the device to be created and enter its MAC address, IP address and firmware version (optional) in the corresponding field.
2. Basic settings will show a list of settings based on the model selected in step 1. Users could assign extensions to accounts, assign functions to Line Keys and Multiple-Purposed Keys if supported on the selected model.
3. Click on "save" to save the configuration for this device.
Figure 68: Create New Device
Manage Devices
The device manually created or discovered from Auto Discover will be listed in the Web GUIValue-added FeaturesZero ConfigZero Config page. Users can see the devices with their MAC address, IP address, vendor, model etc.
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Click on
Figure 69: Manage Devices
to access the Web GUI of the phone.
Click on
to edit the device configuration.
A new dialog will be displayed for the users to configure "Basic" settings and "Advanced" settings. "Basic" settings have the same configurations as displayed when manually creating a new device, i.e., account, line key and MPK settings; "Advanced" settings allow users to configure more details in a five-level structure.
Figure 70: Edit Device
A preview of the "Advanced" settings is shown in the above figure. There are five levels configurations as described in (1) (2) (3) (4) (5) below, with priority from the lowest to the highest. The configurations in all levels will take effect for the device. If there are same options existing in different level configurations with different value configured, the higher-level configuration will override the lower-level configuration.
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(1) Global Policy This is the lowest level configuration. The global policy configured in Web GUIValue-added FeaturesZero ConfigGlobal Policy will be applied here. Clicking on "Modify Global Policy" to redirect to page Value-added FeaturesZero ConfigGlobal Policy.
(2) Global Templates Select a global template to be used for the device and click on
to add. Multiple global templates
can be selected, and users can arrange the priority by adjusting orders via and . All the selected global templates will take effect. If the same option exists on multiple selected global templates, the value in the template with higher priority will override the one in the template with lower priority. Click on
to remove the global template from the selected list. (3) Default Model Template
Default Model Template will be applied to the devices of this model. Default model template can be configured in model template under Web GUIValue-added FeaturesZero ConfigModel Templates page. Please see default model template option in [Table 34: Create New Model Template].
(4) Model Templates
Select a model template to be used for the device and click on to add. Multiple model templates can be selected, and users can arrange the priority by adjusting orders via and . All the selected model templates will take effect. If the same option exists on multiple selected model templates, the value in the template with higher priority will override the one in the template with lower priority. Click on
to remove the model template from the selected list.
(5) Customize Device Settings This is the highest-level configuration for the device. Click on "Modify Customize Device Settings" and following dialog will show.
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Figure 71: Edit Customize Device Settings
Scroll down in the dialog to view and edit the device-specific options. If the users would like to add more options which are not in the pre-defined list, click on "Add New Field" to add a P value number and the value to the configuration. The above figure shows setting P value "P1362" to "en", which means the display language on the LCD is set to English. The warning information on right tells that the option matching the P value number exists and clicking on it will lead to the matching option. For P value information of different models, please refer to configuration template here http://www.grandstream.com/sites/default/files/Resources/config-template.zip.
Select multiple devices that need to be modified and then click on "Update Config" to batch modify devices.
If selected devices are of the same model, the configuration dialog is like the following figure. Configurations in five levels are all available for users to modify.
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Figure 72: Modify Selected Devices - Same Model
If selected devices are of different models, the configuration dialog is like the following figure. Click on to view more devices of other models. Users are only allowed to make modifications in Global Templates and Global Policy level.
Figure 73: Modify Selected Devices - Different Models
---------------------------------------------------------------------------------------------------------------------------------------------------
Note: Performing batch operation will override all the existing device configuration on the page. ---------------------------------------------------------------------------------------------------------------------------------------------------
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After the above configurations, save the changes and go back to Web GUIValue-added FeaturesZero ConfigZero Config page. Users could then click on to send NOTIFY to the SIP end point device and trigger the provisioning process. The device will start downloading the generated configuration file from the URL contained in the NOTIFY message.
Figure 74: Device List in Zero Config
In this web page, users can also click on "Reset All Extensions" to reset the extensions of all the devices.
Sample Application
Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1 GXV3275 phone used by customer support supervisor. 3 of the 8 customer support members speak Spanish and the rest speak English. We could deploy the following configurations to provisioning the office phones for the customer support team.
1. Go to Web GUIValue-added FeaturesZero ConfigZero Config Settings, select "Enable Zero Config".
2. Go to Web GUIValue-added FeaturesZero ConfigGlobal Policy, configure Date Format, Time Format and Firmware Source as follows.
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Figure 75: Zero Config Sample - Global Policy
3. Go to Web GUIValue-added FeaturesZero ConfigModel Templates, create a new model template "English Support Template" for GXP2170. Add option "Language" and set it to "English". Then select the option "Default Model Template" to make it the default model template.
4. Go to Web GUIValue-added FeaturesZero ConfigModel Templates, create another model template "Spanish Support Template" for GXP2170. Add option "Language" and set it to "Español".
5. After 9 devices are powered up and connected to the LAN network, use "Auto Discover" function or "Create New Device" function to add the devices to the device list on Web GUIValue-added FeaturesZero ConfigZero Config.
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6. On Web GUIValue-added FeaturesZero ConfigZero Config page, users could identify the devices
by their MAC addresses or IP addresses displayed on the list. Click on to edit the device settings.
7. For each of the 5 phones used by English speaking customer support, in "Basic settings" select an available extension for account 1 and click on "Save". Then click on "Advanced settings" tab to bring up the following dialog. Users will see the English support template is applied since this is the default model template. A preview of the device settings will be listed on the right side.
Figure 76: Zero Config Sample - Device Preview 1
8. For the 3 phones used by Spanish support, in "Basic settings" select an available extension for account 1 and click on "Save". Then click on "Advanced settings" tab to bring up the following dialog.
Figure 77: Zero Config Sample - Device Preview 2
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Select "Spanish Support Template" in "Model Template". The preview of the device settings is displayed on the
right side and we can see the language is set to "Español" since Model Template has the higher priority for the option "Language", which overrides the value configured in default model template.
9. For the GXV3275 used by the customer support supervisor, select an available extension for account 1 on "Basic settings" and click on "Save". Users can see the preview of the device configuration in "Advanced settings". There is no model template configured for GXV3275.
Figure 78: Zero Config Sample - Device Preview 3
10. Click on "Apply Changes" to apply saved changes.
11. On the Web GUIValue-added FeaturesZero ConfigZero Config page, click on to trigger the device to download config file from UCM630X.
to send NOTIFY
Now all the 9 phones in the network will be provisioned with a unique extension registered on the UCM630X. 3 of the phones will be provisioned to display Spanish on LCD and the other 5 will be provisioned to display English on LCD. The GXV3275 used by the supervisor will be provisioned to use the default language on LCD display since it is not specified in the global policy.
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EXTENSIONS
Create New User Create New SIP Extension
To manually create new SIP user, go to Web GUIExtension/TrunkExtensions. Click on "Add" and a new window will show for users to fill in the extension information.
Figure 79: Create New Device
Extension options are divided into four categories:
Basic Settings Media Features Specific Time Follow me
Select first which type of extension: SIP Extension, IAX Extension or FXS Extension. The configuration parameters are as follows.
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General Extension
CallerID Number
Privilege
SIP/IAX Password Auth ID
Voicemail
Voicemail Password Skip Voicemail Password Verification Send Voicemail Email Notification Attach Voicemail to Email
Table 35: SIP Extension Configuration ParametersBasic Settings
The extension number associated with the user. Configure the CallerID Number that would be applied for outbound calls from this user. Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider. Assign permission level to the user. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". Note: Users need to have the same level as or higher level than an outbound rule's privilege to make outbound calls using this rule. Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this password for security purpose. Configure the authentication ID for the user. If not configured, the extension number will be used for authentication. Configure Voicemail. There are three valid options, and the default option is "Enable Local Voicemail".
Disable Voicemail: Disable Voicemail. Enable Local Voicemail: Enable voicemail for the user. Enable Remote Voicemail: Forward the notify message from remote
voicemail system for the user, and the local voicemail will be disabled. Note: Remote voicemail feature is used only for Infomatec (Brazil). Configure voicemail password (digits only) for the user to access the voicemail box. A random numeric password is automatically generated. It is recommended to use the random generated password for security purpose.
When user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
Configures whether or not to send emails to the extension's email address to notify of new voicemail. Configures whether or not to attach voicemail audio file to the voicemail notification emails. Note: When set to "Default", the global settings in Call Features Voicemail Voicemail Email Settings will be used.
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Keep Voicemail after Emailing
Enable Keep-alive Keep-alive Frequency Enable SCA Emergency CID Name Disable This Extension User Settings First Name Last Name Email Address User Password
Language
Concurrent Registrations Mobile Phone Number
Whether to keep the local voicemail recording after sending them. If set to "Default", the global settings will be used. Note: When set to "Default", the global settings in Call Features Voicemail Voicemail Email Settings will be used. If enabled, empty SDP packet will be sent to the SIP server periodically to keep the NAT port open. The default setting is "No". Configure the Keep-alive interval (in seconds) to check if the host is up. The default setting is 60 seconds. If enabled, (1) Call Forward, Call Waiting and Do Not Disturb settings will not work, (2) Concurrent Registrations can be set only to 1, and (3) Private numbers can be added in Call Features->SCA page.
CallerID name that will be used for emergency calls and callbacks.
If selected, this extension will be disabled on the UCM630X. Note: The disabled extension still exists on the PBX but cannot be used on the end device.
Configure the first name of the user. The first name can contain characters, letters, digits and _. Configure the last name of the user. The last name can contain characters, letters, digits and _. Fill in the Email address for the user. Voicemail will be sent to this Email address. Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purpose. Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under Web GUIPBX SettingsVoice PromptLanguage Settings. The dropdown list shows all the current available voice prompt languages on the UCM630X. To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under Web GUIPBX SettingsVoice PromptLanguage Settings. The maximum endpoints which can be registered into this extension. For security concerns, the default value is 1. Configure the phone number for the extension, user can type the related star code for phone number followed by the extension number to directly call this number. Example: user can type *881000 to call the mobile number associated with extension 1000.
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SIP Settings NAT
Enable Direct Media
DTMF Mode
TEL URI
Alert-Info Enable T.38 UDPTL SRTP
Jitter Buffer
Packet Loss Retransmission Video FEC Audio FEC FECC
Table 36: SIP Extension Configuration ParametersMedia
Use NAT when the UCM630X is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it is related to NAT configuration or Firewall's support of SIP and RTP ports. The default setting is enabled. By default, the UCM630X will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the endpoints to route the media stream directly. It is not always possible for the UCM630X to negotiate endpoint-to-endpoint media routing. The default setting is "No". Select DTMF mode for the user to send DTMF. The default setting is "RFC4733". If "Info" is selected, SIP INFO message will be used. If "Inband" is selected, a-law or ulaw are required. When "Auto" is selected, RFC4733 will be used if offered, otherwise "Inband" will be used. If the phone has an assigned PSTN telephone number, this field should be set to "User=Phone". "User=Phone" parameter will be attached to the Request-Line and "TO" header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel" will be used instead of "SIP" in the SIP request. When present in an INVITE request, the alert-Info header field specifies and alternative ring tone to the UAS.
Enable or disable T.38 UDPTL support.
Enable SRTP for the call. The default setting is disabled. Select jitter buffer method.
Disable: Jitter buffer will not be used. Fixed: Jitter buffer with a fixed size (equal to the value of "jitter buffer size") Adaptive: Jitter buffer with an adaptive size (no more than the value of "max
jitter buffer"). NetEQ: Dynamic jitter buffer via NetEQ. Configure to enable Packet Loss Retransmission. NACK NACK+RTX(SSRC-GROUP) OFF Check to enable Forward Error Correction (FEC) for Video. Check to enable Forward Error Correction (FEC) for Audio. Configure to enable Remote Camera Management.
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ACL Policy Codec Preference
Access Control List manages the IP addresses that can register to this extension. Allow All: Any IP address can register to this extension. Local Network: Only IP addresses in the configured network segments can register to this extension. Press "Add Local Network Address" to add more IP segments.
Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.265, H.263, H.263p, RTX and VP8.
Call Transfer Presence Status
Call Forward Unconditional
CFU Time Condition
Table 37: SIP Extension Configuration ParametersFeatures
Select which presence status to set for the extension and configure call forward conditions for each status. Six possible options are possible: "Available", "Away", "Chat", "Custom", "DND" and "Unavailable". More details at [PRESENCE]. Enable and configure the Call Forward Unconditional target number. Available options for target number are:
"None": Call forward deactivated. "Extension": Select an extension from dropdown list as CFU target. "Custom Number": Enter a customer number as target. For example: *97. "Voicemail": Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension. "Ring Group": Select a ring group from dropdown list as CFU target. "Queues": Select a queue from dropdown list as CFU target. "Voicemail Group": Select a voicemail group from dropdown list as CFU target. The default setting is "None". Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Note: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured under Specific Time section. Scroll down the add
Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
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Call Forward No Answer
Configure the Call Forward No Answer target number. Available options for target number are:
"None": Call forward deactivated. "Extension": Select an extension from dropdown list as CFN target. "Custom Number": Enter a customer number as target. For example: *97. "Voicemail": Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension. "Ring Group": Select a ring group from dropdown list as CFN target. "Queues": Select a queue from dropdown list as CFN target. "Voicemail Group": Select a voicemail group from dropdown list as CFN target. The default setting is "None".
Select time condition for Call Forward No Answer. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific".
CFN Time Condition
Notes:
"Specific" has higher priority to "Office Times" if there is a conflict in terms of time period.
Specific time can be configured under Specific Time section. Scroll down the add Time Condition for specific time.
Office Time and Holiday could be configured on page System SettingsTime SettingsOffice Time/Holiday page.
Call Forward Busy
Configure the Call Forward Busy target number. Available options for target number are:
"None": Call forward deactivated. "Extension": Select an extension from dropdown list as CFB target. "Custom Number": Enter a customer number as target. For example: *97. "Voicemail": Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension. "Ring Group": Select a ring group from dropdown list as CFB target. "Queues": Select a queue from dropdown list as CFB target. "Voicemail Group": Select a voicemail group from dropdown list as CFB target. The default setting is "None".
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CFB Time Condition
Do Not Disturb
DND Time Condition
DND Whitelist
FWD Whitelist
Select time condition for Call Forward Busy. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific".
Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured under Specific Time section. Scroll down the add
Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page. If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored. Select time condition for Do Not Disturb. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific".
Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured under Specific Time section. Scroll down the add
Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime SettingsOffice Time/Holiday page. If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.
Z match any digit from 1-9. N match any digit from 2-9. X match any digit from 0-9. Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported. Z match any digit from 1-9 N match any digit from 2-9 X match any digit from 0-9.
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CC Settings Enable CC
If enabled, UCM630X will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. By default, it is disabled. Two modes for Call Completion are supported:
CC Mode
Normal: This extension is used as ordinary extension. For Trunk: This extension is registered from a PBX.
The default setting is "Normal".
CC Max Agents
Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel can make. The minimum value is 1.
CC Max Monitors
Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.
Ring Simultaneously
Ring Simultaneously
Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.
External Number
Set the external number to be rang simultaneously. `-' is the connection character which will be ignored. This field accepts only letters, numbers, and special characters + = * #.
Time Condition for Ring Simultaneously
Ring the external number simultaneously along with the extension on the basis of this time condition.
Use callee DOD on Use the DOD number when calls are being diverted/forwarded to external
FWD or RS
destinations or when ring simultaneous is configured.
Monitor privilege control
Allowed to call-barging
Add members from "Available Extensions" to "Selected Extensions" so that the selected extensions can spy on the used extension using feature code.
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Seamless transfer privilege control
Allowed to seamless transfer
Any extensions on the UCM can perform seamless transfer. When using Pickup Incall feature, only extensions available on the "Selected Extensions" list can perform seamless transfer to the edited extension.
PMS Remote Wakeup Whitelist
Select the extensions that can set wakeup service for other extensions
Selected extensions can set a PMS wakeup service for this extension via feature code.
Other Settings
Ring Timeout
Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X. The valid range is between 5 seconds and 600 seconds. Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under Web GUICDRRecording Files.
Skip Trunk Auth
If set to "yes", users can skip entering the password when making outbound calls. If set to "By Time", users can skip entering the password when making outbound
calls during the selected time condition. If set to "No", users will be asked to enter the password when making outbound
calls.
Time Condition for If `Skip Trunk Auth' is set to `By Time', select a time condition during which users can Skip Trunk Auth skip entering password when making outbound calls.
Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Support HotDesking Mode
Check to enable Hot-Desking Mode on the extension. Hot-Desking allows to use the same endpoint device and login using extension/password combination. This feature is used in scenarios where different users need to use the same endpoint device during different time of a day for instance. If enabled, SIP Password will accept only alphabet characters and digits. Auth ID will be changed to the same as Extension.
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Enable LDAP
If enabled, the extension will be added to LDAP Phonebook PBX list. Default is enabled.
Use MOH as IVR ringback tone
If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as ringback tone instead of the regular ringback tone.
Music On Hold
Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.
Enable WebRTC Support
Enable registration and call from WebRTC. For more details about WebRTC feature refer to [Wave WebRTC Video Calling & Conferencing]
Bind PMS Room
If enabled, the system will create a room whose room number, by default, will equal the extension number in PMS module. Note: If this room already exists, the configuration of the existing room will be overwritten.
Call Duration Limit Check to enable and set the call limit the duration.
Maximum Call Duration (s)
The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds
The Maximum Number of Call Lines
The maximum number of simultaneous calls that the extension can have. 0 indicates no limit.
Enable AutoAnswer Support
If enabled, the extension will support auto-answer when indicated by Call-info/Alertinfo headers.
Email Missed Calls Send a log of missed calls to the extension's email address.
Missed Call Type Call Waiting
If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:
Default: All missed calls will be sent in email notifications. Missed Internal Call: Only missed local extension-to-extension calls will be
sent in email notifications. Missed External Call: Only missed calls from trunks will be sent in email
notifications.
Allows calls to the extension even when it is already in a call. This only works if the caller is directly dialing the extension. If disabled, the CC service will take effect only for unanswered and timeout calls.
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Specific Time Time Condition
Table 38: SIP Extension Configuration ParametersSpecific Time
Click to add Time Condition to configure specific time for this extension.
Follow Me Enable
Skip Trunk Auth
Music On Hold Class Enable Destination Default Destination Confirm When Answering Use Callee DOD for Follow Me
New Follow Me Number
Dialing Order
Table 39: Table 34: SIP Extension Configuration ParametersFollow Me
Configure to enable or disable Follow Me for this user. If the outbound calls need to check the password, we should enable this option or enable the option "Skip Trunk Auth" of the Extension. Otherwise this Follow Me cannot call out. Configure the Music On Hold class that the caller would hear while tracking the user.
Configure to enable destination.
The call will be routed to this destination if no one in the Follow Me answers the call.
If enabled, call will need to be confirmed after answering. Use the callee DOD number as CID if configured Follow Me numbers are external numbers. Add a new Follow Me number which could be a "Local Extension" or an "External Number". The selected dial plan should have permissions to dial the defined external number. This is the order in which the Follow Me destinations will be dialed to reach the user.
Create New IAX Extension
The UCM630X supports Inter-Asterisk eXchange (IAX) protocol. IAX is used for transporting VoIP telephony sessions between servers and terminal devices. IAX is like SIP but also has its own characteristic. For more information, please refer to RFC 5465.
To manually create new IAX user, go to Web GUIExtension/TrunkExtensions. Click on "Add" and a new dialog window will show for users which need to make sure first to select the extension type to be IAX Extension before proceeding to fill in the extension information. The configuration parameters are as follows.
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Table 40: IAX Extension Configuration ParametersBasic Settings
General
Extension
The extension number associated with the user.
CallerID Number
Configure the CallerID Number that would be applied for outbound calls from this user. Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider.
Privilege
Assign permission level to the user. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". Note: Users need to have the same level as or higher level than an outbound rule's privilege to make outbound calls using this rule.
Configure the password for the user. A random secure password will be automatically SIP/IAX Password
generated. It is recommended to use this password for security purpose.
Voicemail
Configure Voicemail. There are three valid options and the default option is "Enable Local Voicemail".
Disable Voicemail: Disable Voicemail. Enable Local Voicemail: Enable voicemail for the user.
Voicemail Password
Configure voicemail password (digits only) for the user to access the voicemail box. A random numeric password is automatically generated. It is recommended to use the random generated password for security purpose.
Skip Voicemail Password Verification
When user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
Send Voicemail
Configures whether or not to send emails to the extension's email address to notify of
Email Notification new voicemail.
Attach Voicemail to Email
Configures whether or not to attach voicemail audio file to the voicemail notification emails.
Keep Voicemail after Emailing
Only applies if extension-level or global Send Voicemail to Email is enabled.
Disable This Extension
If selected, this extension will be disabled on the UCM630X.
Note: The disabled extension still exists on the PBX but cannot be used on the end device.
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User Settings First Name Last Name Email Address User Password
Language
Mobile Phone Number
Configure the first name of the user. The first name can contain characters, letters, digits and _. Configure the last name of the user. The last name can contain characters, letters, digits and _. Fill in the Email address for the user. Voicemail will be sent to this Email address. Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purpose. Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under Web GUIPBX SettingsVoice PromptLanguage Settings. The dropdown list shows all the current available voice prompt languages on the UCM630X. To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under Web GUIPBX SettingsVoice PromptLanguage Settings.
Configure the Mobile number of the user.
IAX Settings Max Number of Calls Require Call Token SRTP
ACL Policy
Codec Preference
Table 41: IAX Extension Configuration ParametersMedia
Configure the maximum number of calls allowed for each remote IP address.
Configure to enable/disable requiring call token. If set to "Auto", it might lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints. The default setting is "Yes". Enable SRTP for the call. The default setting is disabled. Access Control List manages the IP addresses that can register to this extension. Allow All: Any IP address can register to this extension. Local Network: Only IP addresses in the configured network segments can
register to this extension. Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.265, H.263, H.263p, RTX and VP8.
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Table 42: IAX Extension Configuration ParametersFeatures
Call Transfer
Call Forward Unconditional
Configure the Call Forward Unconditional target number. If not configured, the Call Forward Unconditional feature is deactivated. The default setting is deactivated.
CFU Time Condition
Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Note: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Call Forward No Answer
Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is deactivated. The default setting is deactivated.
CFN Time Condition
Select time condition for Call Forward No Answer. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Configure the Call Forward Busy target number. If not configured, the Call Forward Call Forward Busy
Busy feature is deactivated. The default setting is deactivated.
CFB Time Condition
Select time condition for Call Forward Busy. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period.
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Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.
Office Time and Holiday could be configured on page System SettingsTime SettingsOffice Time/Holiday page.
Do Not Disturb
If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.
DND Time Condition
The time condition of DND. The DND will take effect while the time condition is satisfied.
DND Whitelist
If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.
Z match any digit from 1-9. N match any digit from 2-9. X match any digit from 0-9.
FWD Whitelist
Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.
Z match any digit from 1-9. N match any digit from 2-9. X match any digit from 0-9.
Ring Simultaneously
Ring Simultaneously
Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.
External Number
Set the external number to be rang simultaneously. `-' is the connection character which will be ignored.
Time Condition for Ring Simultaneously
Ring the external number simultaneously along with the extension on the basis of this time condition.
Use callee DOD on Use the callee's DOD number as CallerID on Outgoing Forwarding or Ring
FWD or RS
Simultaneously calls.
Monitor privilege control
Allow call-barging Members of the list can spy on this extension via feature codes.
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Seamless transfer privilege control
Allowed to Members of the list can seamlessly transfer via feature code.
seamless transfer
Other Settings
Ring Timeout
Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X, which can be configured in the global ring timeout setting under Web GUIPBX SettingsVoice PromptCustom Prompt: General Preference. The valid range is between 5 seconds and 600 seconds. Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUICDRRecording Files.
Skip Trunk Auth
If set to "Yes", users can skip entering the password when making outbound calls. If set to "By Time", users can skip entering the password when making outbound
calls during the selected time condition.
If set to "No", users will be asked to enter the password when making outbound calls.
Time Condition for If "Skip Trunk Auth" is set to "By Time", select a time condition during which users can Skip Trunk Auth skip entering password when making outbound calls.
Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Enable LDAP
If enabled, the extension will be added to LDAP Phonebook PBX lists.
Music On Hold
Configure the Music On Hold class to suggest to the bridged channel when putting them on hold.
Use MOH as IVR ringback tone
If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as ringback tone instead of the regular ringback tone.
Call Duration Limit Check to enable and set the call limit the duration.
Maximum Call Duration (s)
The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds
Email Missed
Send a log of missed calls to the extension's email address.
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Calls Missed Call Type
If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:
Default: All missed calls will be sent in email notifications. Missed Internal Call: Only missed local extension-to-extension calls will be
sent in email notifications. Missed External Call: Only missed calls from trunks will be sent in email
notifications.
Specific Time Time Condition
Table 43: IAX Extension Configuration ParametersSpecific Time
Click to add Time Condition to configure specific time for this extension.
Follow Me Enable
Skip Trunk Auth
Music On Hold Class Enable Destination Default Destination Confirm When Answering Use Callee DOD for Follow Me
New Follow Me Number
Dialing Order
Table 44: IAX Extension Configuration ParametersFollow Me
Configure to enable or disable Follow Me for this user. If the outbound calls need to check the password, we should enable this option or enable the option "Skip Trunk Auth" of the Extension. Otherwise this Follow Me cannot call out. Configure the Music On Hold class that the caller would hear while tracking the user.
Configure to enable destination.
The call will be routed to this destination if no one in the Follow Me answers the call.
If enabled, call will need to be confirmed after answering. Use the callee DOD number as CID if configured Follow Me numbers are external numbers. Add a new Follow Me number which could be a "Local Extension" or an "External Number". The selected dial plan should have permissions to dial the defined external number. This is the order in which the Follow Me destinations will be dialed to reach the user.
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Create New FXS Extension
The UCM630X supports Foreign eXchange Subscriber (FXS) interface. FXS is used when user needs to connect analog phone lines or FAX machines to the UCM630X.
To manually create new FXS user, go to Web GUIExtension/TrunkExtensions. Click on "Add" and a new dialog window will show for users which need to make sure first to select the extension type to be FXS Extension before proceeding to fill in the extension information. The configuration parameters are as follows.
General Extension Analog Station
Caller ID Number
Privilege
Voicemail
Voicemail Password Skip Voicemail Password Verification Send Voicemail Email Notification Attach Voicemail to Email Keep Voicemail
Table 45: FXS Extension Configuration ParametersBasic Settings
The extension number associated with the user. Select the FXS port to be assigned for this extension. Configure the CallerID Number that would be applied for outbound calls from this user. Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP provider. Assign permission level to the user. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". Note: Users need to have the same level as or higher level than an outbound rule's privilege to make outbound calls using this rule. Configure Voicemail. There are three valid options and the default option is "Enable Local Voicemail".
Disable Voicemail: Disable Voicemail. Enable Local Voicemail: Enable voicemail for the user. Configure voicemail password (digits only) for the user to access the voicemail box. A random numeric password is automatically generated. It is recommended to use the random generated password for security purpose.
When user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
Configures whether or not to send emails to the extension's email address to notify of new voicemail. Configures whether or not to attach voicemail audio file to the voicemail notification emails. Only applies if extension-level or global Send Voicemail to Email is enabled.
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after Emailing Emergency CID Name Disable This Extension User Settings First Name
Last Name Email Address
User Password
Mobile Phone Number
Language
CallerID name that will be used for emergency calls and callbacks.
If selected, this extension will be disabled on the UCM630X. Note: The disabled extension still exists on the PBX but cannot be used on the end device.
Configure the first name of the user. The first name can contain characters, letters, digits and _. Configure the last name of the user. The last name can contain characters, letters, digits and _. Fill in the Email address for the user. Voicemail will be sent to this Email address. Configure the password for user portal access. A random numeric password is automatically generated. It is recommended to use the randomly generated password for security purpose.
Configure the Mobile number of the user.
Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under Web GUIPBX SettingsVoice PromptLanguage Settings. The dropdown list shows all the current available voice prompt languages on the UCM630X. To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under Web GUIPBX SettingsVoice PromptLanguage Settings.
Analog Settings Call Waiting User `#' as SEND RX Gain TX Gain
Min RX Flash
Table 46: FXS Extension Configuration ParametersMedia
Configure to enable/disable call waiting feature. The default setting is "No". If configured, the # key can be used as SNED key after dialing the number on the analog phone. The default setting is "Yes". Configure the RX gain for the receiving channel of analog FXS port. The valid range is -30dB to +6dB. The default setting is 0. Configure the TX gain for the transmitting channel of analog FXS port. The valid range is -30dB to +6dB. The default setting is 0. Configure the minimum period of time (in milliseconds) that the hook-flash must remain unpressed for the PBX to consider the event as a valid flash event. The valid range is 30ms to 1000ms. The default setting is 200ms.
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Max RX Flash
Enable Polarity Reversal
Echo Cancellation
3-Way Calling Send CallerID After
Fax Mode
Configure the maximum period of time (in milliseconds) that the hook-flash must remain unpressed for the PBX to consider the event as a valid flash event. The minimum period of time is 256ms and it cannot be modified. The default setting is 1250ms. If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as Hangup on a polarity reversal. The default setting is "Yes". Specify "ON", "OFF" or a value (the power of 2) from 32 to 1024 as the number of taps of cancellation. Note: When configuring the number of taps, the number 256 is not translated into 256ms of echo cancellation. Instead, 256 taps mean 256/8 = 32 ms. The default setting is "ON", which is 128 taps. Configure to enable/disable 3-way calling feature on the user. The default setting is enabled.
Configure the number of rings before sending CID. Default setting is 1.
For FXS extension, there are three options available in Fax Mode. The default setting is "None". None: Disable Fax. Fax Gateway: If selected, the UCM630X can support conversation and
processing of Fax data from T.30 to T.38 or T.38 to T.30. only for FXS ports. Fax Detection: During a call, the fax signal from the user/trunk will be detected,
and the received fax will be sent to the email address configured for the user. If an email address has been configured for the user, the fax will be sent to the Default Email Address configured in Fax/T.38->Fax Settings .
Call Transfer Call Forward Unconditional
CFU Time Condition
Table 47: FXS Extension Configuration ParametersFeatures
Configure the Call Forward Unconditional target number. If not configured, the Call Forward Unconditional feature is deactivated. The default setting is deactivated. Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Note: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
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Call Forward No Answer
CFN Time Condition
Call Forward Busy
Office Time and Holiday could be configured on page System SettingsTime SettingsOffice Time/Holiday page.
Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is deactivated. The default setting is deactivated.
Select time condition for Call Forward No Answer. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Configure the Call Forward Busy target number. If not configured, the Call Forward Busy feature is deactivated. The default setting is deactivated.
CFB Time Condition
Select time condition for Call Forward Busy. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Do Not Disturb
If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.
DND Time Condition
The time condition of DND. The DND will take effect while the time condition is satisfied.
DND Whitelist
If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.
Z match any digit from 1-9. N match any digit from 2-9. X match any digit from 0-9..
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Calls from users in the forward whitelist will not be forwarded.
FWD Whitelist
Pattern matching is supported. Z match any digit from 1-9. N match any digit from 2-9. X match any digit from 0-9.
CC Settings
Enable CC
If enabled, UCM630X will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason.
Ring Simultaneously
Ring Simultaneously
Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.
External Number
Set the external number to be rang simultaneously. `-' is the connection character which will be ignored.
Time Condition for Ring Simultaneously
Ring the external number simultaneously along with the extension on the basis of this time condition.
Use callee DOD on Use the callee's DOD number as CallerID on Outgoing Forwarding or Ring
FWD or RS
Simultaneously calls.
Hotline
Enable Hotline
If enabled, hotline dialing plan will be activated, a pre-configured number will be used according to the selected Hotline Type.
Hotline Number
Configure the Hotline Number
Configure the Hotline Type:
Hotline Type
Immediate Hotline: When the phone is off-hook, UCM630X will immediately dial the preset number
Delay Hotline: When the phone is off hook, if there is no dialing within 5 seconds, UCM630X will
dial the preset number.
Monitor privilege control
Allow call-barging Members of the list can spy on this extension via feature codes.
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Seamless transfer privilege control
Allowed to Members of the list can seamlessly transfer via feature code.
seamless transfer
Other Settings
Ring Timeout
Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X, which can be configured in the global ring timeout setting under Web GUIPBX SettingsVoice PromptCustom Prompt: General Preference. The valid range is between 5 seconds and 600 seconds. Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUICDRRecording Files.
Skip Trunk Auth
If set to "Yes", users can skip entering the password when making outbound calls. If set to "By Time", users can skip entering the password when making outbound
calls during the selected time condition. If set to "No", users will be asked to enter the password when making outbound
calls.
Time Condition for If "Skip Trunk Auth" is set to "By Time", select a time condition during which users can Skip Trunk Auth skip entering password when making outbound calls.
Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Enable LDAP
If enabled, this extension will be added to LDAP Phonebook PBX list; if disabled, this extension will be skipped when creating LDAP Phonebook.
Use MOH as IVR ringback tone
If enabled, when the call to the extension is made through the IVR, the caller will hear MOH as ringback tone instead of the regular ringback tone.
Music On Hold
Select which Music On Hold class to suggest to extension when putting the active call on hold.
Call Duration Limit Check to enable and set the call limit the duration.
Maximum Call Duration (s)
The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds
Email Missed Calls
Send a log of missed calls to the extension's email address.
Missed Call Type
If Email Missed Calls enabled, users can select the type of missed calls to be sent via email, the available types are:
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Default: All missed calls will be sent in email notifications. Missed Internal Call: Only missed local extension-to-extension calls will be
sent in email notifications. Missed External Call: Only missed calls from trunks will be sent in email
notifications.
Specific Time Time Condition
Table 48: FXS Extension Configuration ParametersSpecific Time
Click to add Time Condition to configure specific time for this extension.
Follow Me Enable
Skip Trunk Auth
Music On Hold Class Enable Destination Default Destination Confirm When Answering Use Callee DOD for Follow Me
New Follow Me Number
Dialing Order
Table 49: FXS Extension Configuration ParametersFollow Me
Configure to enable or disable Follow Me for this user. If the outbound calls need to check the password, we should enable this option or enable the option "Skip Trunk Auth" of the Extension. Otherwise this Follow Me cannot call out. Configure the Music On Hold class that the caller would hear while tracking the user.
Configure to enable destination.
The call will be routed to this destination if no one in the Follow Me answers the call.
If enabled, call will need to be confirmed after answering. Use the callee DOD number as CID if configured Follow Me numbers are external numbers. Add a new Follow Me number which could be a "Local Extension" or an "External Number". The selected dial plan should have permissions to dial the defined external number. This is the order in which the Follow Me destinations will be dialed to reach the user.
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Batch Add Extensions
Batch Add SIP Extensions
To add multiple SIP extensions, BATCH add can be used to create standardized SIP extension accounts. However, unique extension username cannot be set using BATCH add. Under Web GUIExtension/TrunkExtensions, click on "Add" and select extension type as SIP extension, and "Select Add Method" as Batch.
General Create Number Extension Incrementation Extension Permission
Voicemail
SIP/IAX Password
Voicemail Password
Table 50: Batch Add SIP Extension Parameters
Specify the number of extensions to be added. The default setting is 5. Select how much to increment successive extensions. For example, if the value is 2, the extensions will be 1000,1002,1004,...... Note: Up to 3 characters.
Configure the starting extension number of the batch of extensions to be added.
Assign permission level to the user. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". Note: Users need to have the same level as or higher level than an outbound rule's privilege to make outbound calls from this rule. Configure Voicemail. There are three valid options and the default option is "Enable Local Voicemail".
Disable Voicemail: Disable Voicemail. Enable Local Voicemail: Enable voicemail for the user. Enable Remote Voicemail: Forward the notify message from remote
voicemail system for the user, and the local voicemail will be disabled. Note: Remote voicemail feature is used only for Infomatec (Brazil). Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions. User Random Password. A random secure password will be automatically generated. It is recommended to use this password for security purpose. Use Extension as Password. Enter a password to be used on all the extensions in the batch. Configure Voicemail password (digits only) for the users. User Random Password. A random password in digits will be automatically generated. It is recommended to use this password for security purpose. Use Extension as Password.
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Send Voicemail to Email
Keep Voicemail after Emailing
Enter a password to be used on all the extensions in the batch. Send voicemail messages to the configured email address. If set to "Default", the global setting will be used. Global settings can be found in Voicemail->Voicemail Email Settings.
Only applies if extension-level or global Send Voicemail to Email is enabled.
Configure CallerID Number when adding Batch Extensions.
CallerID Number
Use Extension as Number Users can choose to use the extension number as CallerID
Skip Voicemail Password Verification Enable Keep-alive Keep-alive Frequency Disable This Extension
Enable SCA
Emergency Calls CID
Language
Media
Use as Number Users can choose to set a specific number instead of using the extension number.
When user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
If enabled, the PBX will regularly send SIP OPTIONS to check if host device is online.
Configure the keep-alive interval (in seconds) to check if the host is up.
Check this box to disable this extension.
If enabled, (1) Call Forward, Call Waiting and Do Not Disturb settings will not work, (2) Concurrent Registrations can be set only to 1, and (3) Private numbers can be added in Call Features->SCA page.
CallerID number that will be used when calling out and receiving direct callbacks.
Select voice prompt language for this extension. If set to "Default", the global setting for voice prompt language will be used.
Use NAT when the PBX is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it is related to NAT NAT configuration or Firewall's support of SIP and RTP ports. The default setting is enabled.
Enable Direct Media
By default, the PBX will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the endpoints to route the media stream directly. It is not always possible for the PBX to negotiate endpoint-to-endpoint media routing. The default setting is "No".
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DTMF Mode
Select DTMF mode for the user to send DTMF. The default setting is "RFC4733". If "Info" is selected, SIP INFO message will be used. If "Inband" is selected, a-law or ulaw are required. When "Auto" is selected, RFC4733 will be used if offered, otherwise "Inband" will be used.
Alert-info
When present in an INVITE request, the Alert-info header field specifies an alternative ring tone to the UAS.
SRTP
Enable/disable SRTP for RTP stream encryption.
Packet Loss Retransmission
Video FEC
Configure to enable Packet Loss Retransmission. NACK NACK+RTX(SSRC-GROUP) OFF
Check to enable Forward Error Correction (FEC) for Video.
FECC
Configure to enable FECC
Audio FEC
Check to enable Forward Error Correction (FEC) for Audio.
ACL Policy
Jitter Buffer Codec Preference Call Transfer Presence Status
Access Control List manages the IP addresses that can register to this extension. Allow All: Any IP address can register to this extension. Local Network: Only IP addresses in the configured network segments can register to this extension. Press "Add Local Network Address" to add more IP segments.
Select jitter buffer method. Disable: Jitter buffer will not be used. Fixed: Jitter buffer with a fixed size (equal to the value of "jitter buffer size") Adaptive: Jitter buffer with an adaptive size (no more than the value of "max jitter buffer"). NetEQ: Dynamic jitter buffer via NetEQ.
Configure the codecs to be used.
Select which presence status to set for the extension and configure call forward conditions for each status. Six possible options are possible: "Available", "Away", "Chat", "Custom", "DND" and "Unavailable". More details at [PRESENCE].
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Call Forward Unconditional
CFU Time Condition
Call Forward No Answer
CFN Time Condition
Enable and configure the Call Forward Unconditional target number. Available options for target number are:
"None": Call forward deactivated. "Extension": Select an extension from dropdown list as CFU target. "Custom Number": Enter a customer number as target. For example: *97. "Voicemail": Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension. "Ring Group": Select a ring group from dropdown list as CFU target. "Queues": Select a queue from dropdown list as CFU target. "Voicemail Group": Select a voicemail group from dropdown list as CFU target. The default setting is "None".
Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Note: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Configure the Call Forward No Answer target number. Available options for target number are:
"None": Call forward deactivated. "Extension": Select an extension from dropdown list as CFN target. "Custom Number": Enter a customer number as target. For example: *97. "Voicemail": Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension. "Ring Group": Select a ring group from dropdown list as CFN target. "Queues": Select a queue from dropdown list as CFN target. "Voicemail Group": Select a voicemail group from dropdown list as CFN target. The default setting is "None".
Select time condition for Call Forward No Answer. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or
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Holiday" and "Specific".
Notes:
"Specific" has higher priority to "Office Times" if there is a conflict in terms of time period.
Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.
Office Time and Holiday could be configured on page System SettingsTime SettingsOffice Time/Holiday page.
Call Forward Busy
Configure the Call Forward Busy target number. Available options for target number are:
"None": Call forward deactivated. "Extension": Select an extension from dropdown list as CFB target. "Custom Number": Enter a customer number as target. For example: *97. "Voicemail": Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension. "Ring Group": Select a ring group from dropdown list as CFB target. "Queues": Select a queue from dropdown list as CFB target. "Voicemail Group": Select a voicemail group from dropdown list as CFB target. The default setting is "None".
Select time condition for Call Forward Busy. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific".
CFB Time Condition
Do Not Disturb DND Time Condition
Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.
Select time condition for Do Not Disturb. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday"
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DND Whitelist FWD Whitelist CC Settings Enable CC
CC Mode
and "Specific".
Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime SettingsOffice Time/Holiday page. If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported. Z match any digit from 1-9N match any digit from 2-9X match any digit from 0-9.
Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported. Z match any digit from 1-9N match any digit from 2-9X match any digit from 0-9.
If enabled, UCM630X will automatically alert this extension when a called party is available, given that a previous call to that party failed for some reason. By default, it is disabled.
Two modes for Call Completion are supported: Normal: This extension is used as ordinary extension. For Trunk: This extension is registered from a PBX. The default setting is "Normal".
CC Max Agents
Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel can make. The minimum value is 1.
CC Max Monitors
Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.
Ring Simultaneously
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Ring Simultaneously
Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.
External Number
Set the external number to be rang simultaneously. `-' is the connection character which will be ignored. This field accepts only letters, numbers, and special characters + = * #.
Time Condition for Ring Simultaneously
Ring the external number simultaneously along with the extension on the basis of this time condition.
Use callee DOD on Use the DOD number when calls are being diverted/forwarded to external
FWD or RS
destinations or when ring simultaneous is configured.
Monitor privilege control
Allowed to call-barging
Add members from "Available Extensions" to "Selected Extensions" so that the selected extensions can spy on the used extension using feature code.
Seamless transfer privilege control
Allowed to seamless transfer
Any extensions on the UCM can perform seamless transfer. When using Pickup Incall feature, only extensions available on the "Selected Extensions" list can perform seamless transfer to the edited extension.
Other Settings
Ring Timeout
Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X, which can be configured in the global ring timeout setting under Web GUIPBX SettingsVoice PromptCustom Prompt: General Preference. The valid range is between 3 seconds and 600 seconds. Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under Web GUICDRRecording Files.
Skip Trunk Auth
If set to "yes", users can skip entering the password when making outbound calls. If set to "By Time", users can skip entering the password when making outbound
calls during the selected time condition. If set to "No", users will be asked to enter the password when making outbound
calls.
Time Condition for If `Skip Trunk Auth' is set to `By Time', select a time condition during which users can
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Skip Trunk Auth skip entering password when making outbound calls.
Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Enable LDAP
If enabled, the extension will be added to LDAP Phonebook PBX list.
Enable WebRTC Support
Enable registration and call from WebRTC.
Bind PMS Room
If enabled, the system will create a room whose room number, by default, will equal the extension number in PMS module. Note: If this room already exists, the configuration of the existing room will be overwritten.
Music On Hold
Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.
Call Duration Limit The maximum duration of call-blocking.
Maximum Call Duration
The maximum call duration (in seconds). The default value 0 means no limit.
Call Waiting
If disabled, UCM will not invite the extension when it is already in a call and will do the same work as the user is busy.
Note: the option only works when the caller dials the extension directly.
Batch Add IAX Extensions
Under Web GUIExtension/TrunkExtensions, click on "Add", then select extension type as IAX Extension and the add method to be Batch.
T
able 51: Batch Add IAX Extension Parameters
General
Create Number
Specify the number of extensions to be added. The default setting is 5.
Extension Incrementation
Select how much to increment successive extensions. For example, if the value is 2, the extensions will be 1000,1002,1004,......
Extension
The extension number associated with this particular user/phone.
Permission
Assign permission level to the user. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". Note: Users need to have the same level as or higher level than an outbound rule's privilege in order to make outbound calls from this rule.
CallerID Number
Configure the Caller ID number displayed when dialing calls from this user. Note: The Caller ID usage might be limited by your VoIP provider. In Batch Add Method,
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"e" means to use the extension as the number.
Voicemail
Configure Voicemail. There are three valid options and the default option is "Enable Local Voicemail".
Disable Voicemail: Disable Voicemail. Enable Local Voicemail: Enable voicemail for the user.
Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.
SIP/IAX Password
User Random Password. A random secure password will be automatically generated. It is recommended to use this password for security purpose.
Use Extension as Password. Enter a password to be used on all the extensions in the batch.
Configure Voicemail password (digits only) for the users.
Voicemail Password
User Random Password. A random password in digits will be automatically generated. It is recommended to use this password for security purpose.
Use Extension as Password. Enter a password to be used on all the extensions in the batch.
Send Voicemail to Email
Send voicemail messages to the configured email address. If set to "Default", the global setting will be used. Global settings can be found in Voicemail->Voicemail Email Settings.
Keep Voicemail after Emailing
Only applies if extension-level or global Send Voicemail to Email is enabled.
Auto Record
Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under Web GUICDRRecording Files.
Skip Voicemail Password Verification
When user dials voicemail code, the password verification IVR is skipped. If enabled, this would allow one-button voicemail access. By default, this option is disabled.
Disable This Extension
Check this box to disable this extension.
Language
Select voice prompt language for this extension. If set to "Default", the global setting for voice prompt language will be used.
IAX Settings
Max Number of Calls Configure the maximum number of calls allowed for each remote IP address.
Require Call Token
Configure to enable/disable requiring call token. If set to "Auto", it might lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints.
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SRTP ACL Policy Codec Preference
The default setting is "Yes". Enable/disable SRTP for RTP stream encryption. Access Control List manages the IP addresses that can register to this extension.
Allow All: Any IP address can register to this extension. Local Network: Only IP addresses in the configured network segments can
register to this extension. Configure the codecs to be used.
Call Transfer Call Forward Unconditional
CFU Time Condition
Call Forward No Answer
Enable and configure the Call Forward Unconditional target number.
Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific". Note: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page. Configure the Call Forward No Answer target number.
Select time condition for Call Forward No Answer. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific".
CFN Time Condition
Call Forward Busy CFB Time
Notes:
"Specific" has higher priority to "Office Times" if there is a conflict in terms of time period.
Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.
Office Time and Holiday could be configured on page System SettingsTime SettingsOffice Time/Holiday page.
Configure the Call Forward Busy target number. Select time condition for Call Forward Busy. The available time conditions are "Office
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Condition
Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific".
Do Not Disturb
Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
If Do Not Disturb is enabled, all incoming calls will be dropped. All call forward settings will be ignored.
Select time condition for Do Not Disturb. The available time conditions are "Office Time", "Out of Office Time", "Holiday", "Out of Holiday", "Out of Office Time or Holiday" and "Specific".
DND Time Condition
Notes: "Specific" has higher priority to "Office Times" if there is a conflict in terms of time
period. Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time. Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
DND Whitelist
If DND is enabled, calls from the whitelisted numbers will not be rejected. Multiple numbers are supported and must be separated by new lines. Pattern matching is supported.
Z match any digit from 1-9 N match any digit from 2-9 X match any digit from 0-9.
Calls from users in the forward whitelist will not be forwarded. Pattern matching is supported.
FWD Whitelist
Z match any digit from 1-9 N match any digit from 2-9 X match any digit from 0-9.
Ring Simultaneously
Ring Simultaneously
Enable this option to have an external number ring simultaneously along with the extension. If a register trunk is used for outbound, the register number will be used to be displayed for the external number as caller ID number.
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External Number
Set the external number to be rang simultaneously. `-' is the connection character which will be ignored. This field accepts only letters, numbers, and special characters + = * #.
Time Condition for Ring Simultaneously
Ring the external number simultaneously along with the extension on the basis of this time condition.
Use callee DOD on Use the DOD number when calls are being diverted/forwarded to external
FWD or RS
destinations or when ring simultaneous is configured.
Monitor privilege control
Allowed to call-barging
Add members from "Available Extensions" to "Selected Extensions" so that the selected extensions can spy on the used extension using feature code.
Seamless transfer privilege control
Allowed to seamless transfer
Any extensions on the UCM can perform seamless transfer. When using Pickup Incall feature, only extensions available on the "Selected Extensions" list can perform seamless transfer to the edited extension.
Other Settings
Ring Timeout
Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the UCM630X, which can be configured in the global ring timeout setting under Web GUIPBX SettingsVoice PromptCustom Prompt: General Preference. The valid range is between 5 seconds and 600 seconds. Note: If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.
Auto Record
Enable automatic recording for the calls using this extension. The default setting is disabled. The recordings can be accessed under Web GUICDRRecording Files.
Skip Trunk Auth
If set to "yes", users can skip entering the password when making outbound calls. If set to "By Time", users can skip entering the password when making outbound
calls during the selected time condition. If set to "No", users will be asked to enter the password when making outbound
calls.
Time Condition for If `Skip Trunk Auth' is set to `By Time', select a time condition during which users can Skip Trunk Auth skip entering password when making outbound calls.
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Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Enable LDAP
If enabled, the extension will be added to LDAP Phonebook PBX list.
Music On Hold
Specify which Music On Hold class to suggest to the bridged channel when putting them on hold.
Call Duration Limit Check to enable and set the call limit the duration.
Maximum Call Duration (s)
The maximum call duration (in seconds). The default value 0 means no limit. Max value is 86400 seconds
Batch Extension Resetting Functionality
Users can select multiple extensions and reset their settings to default by pressing the reset button and confirm the reset functionality. Once done, all settings in Basic Settings page will be restored to default values with the exception of Concurrent Registrations. User voicemail password will be reset to Random Password. User voicemail prompts and recordings will be deleted. User Management settings will also be restored to default with the exception of usernames and custom privileges
Search and Edit Extension
All the UCM630X extensions are listed under Web GUIExtension/TrunkExtensions, with status, Extension, CallerID Name, Technology (SIP, IAX and FXS), IP and Port. Each extension has a checkbox for users to "Edit" or "Delete". Also, options "Edit" , "Reboot" and "Delete" are available per extension. User can search an extension by specifying the extension number to find an extension quickly.
Figure 80: Manage Extensions
Status
Users can see the following icon for each extension to indicate the SIP status.
Green:
Idle
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Blue: Yellow: Grey:
Ringing In Use Unavailable (the extension is not registered or disabled on the PBX)
Edit single extension
Click on
to start editing the extension parameters.
Reset single extension
Click on to reset the extension parameters to default (except concurrent registration). Other settings will be restored to default in MaintenanceUser ManagementUser Information except username and permissions and delete the user voicemail prompt and voice messages.
Reboot the user
Click on
to send NOTIFY reboot event to the device which has an UCM630X extension already
registered. To successfully reboot the user, "Zero Config" needs to be enabled on the UCM630X Web GUIValue-added FeaturesZero ConfigZero Config Settings.
Delete single extension
Click on
to delete the extension. Or select the checkbox of the extension and then click on "Delete
Selected Extensions".
Modify selected extensions Select the checkbox for the extension(s). Then click on "Edit" to edit the extensions in a batch.
Delete selected extensions Select the checkbox for the extension(s). Then click on "Delete " to delete the extension(s).
Export Extensions
The extensions configured on the UCM630X can be exported to csv format file with selected technology "SIP", "IAX" or "FXS". Click on "Export Extensions" button and select technology in the prompt below.
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Figure 81: Export Extensions
The exported csv file can serve as a template for users to fill in desired extension information to be imported to the UCM630X.
Import Extensions
The capability to import extensions to the UCM630X provides users flexibility to batch add extensions with similar or different configuration quickly into the PBX system.
1. Export extension csv file from the UCM630X by clicking on "Export Extensions" button. 2. Fill up the extension information you would like in the exported csv template. 3. Click on "Import Extensions" button. The following dialog will be prompted.
Figure 82: Import Extensions
4. Select the option in "On Duplicate Extension" to define how the duplicate extension(s) in the imported csv file should be treated by the PBX. Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension information as previously configured without change.
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Delete and Recreate: The current extension previously configured will be deleted and the duplicate extension in the csv file will be loaded to the PBX.
Update Information: The current extension previously configured in the PBX will be kept. However, if the duplicate extension in the csv file has different configuration for any options, it will override the configuration for those options in the extension.
5. Click on "Choose file to upload" to select csv file from local directory in the PC. 6. Click on "Apply Changes" to apply the imported file on the UCM630X.
Example of file to import:
Figure 83: Import File
Table 52: SIP extensions Imported File Example
Field
Supported values
Extension
Digits
Technology
SIP/SIP(WebRTC)
Enable Voicemail
yes/no/remote
CallerID Number
Digits
SIP/IAX Password
Alphanumeric characters
Voicemail Password
Digits
Skip Voicemail Password Verification
yes/no
Ring Timeout
Empty/ 3 to 600 (in second)
SRTP
yes/no
Strategy
Allow All/Local Subnet Only/A Specific IP Address
Local Subnet 1
IP address/Mask
Local Subnet 2
IP address/Mask
Local Subnet 3
IP address/Mask
Local Subnet 4
IP address/Mask
Local Subnet 5
IP address/Mask
Local Subnet 6
IP address/Mask
Local Subnet 7
IP address/Mask
Local Subnet 8
IP address/Mask
Local Subnet 9
IP address/Mask
Local Subnet 10
IP address/Mask
Specific IP Address
IP address
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Skip Trunk Auth Codec Preference
Permission NAT DTMF Mode Insecure Enable Keep-alive Keep-alive Frequency AuthID TEL URI Call Forward Busy Call Forward No Answer Call Forward Unconditional Support Hot-Desking Mode Dial Trunk Password Disable This Extension CFU Time Condition
CFN Time Condition
CFB Time Condition
Music On Hold CC Agent Policy
CC Monitor Policy CCBS Available Timer CCNR Available Timer CC Offer Timer CC Max Agents CC Max Monitors Ring simultaneously External Number
yes/no/bytime PCMU,PCMA,GSM,G.726,G.722,G.729,H.264, H.265,ILBC,AAL2-G.72632,ADPCM,G.723,H.263,H.263p,vp8,opus Internal/Local/National/International yes/no RFC4733/info/inband/auto Port Yes/no Value from 1-3600 Alphanumeric value without special characters Disabled/user=phone/enabled Digits Digits Digits Yes/no Digits Yes/no All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time Default/ringbacktone_default If CC is disabled use: never If CC is set to normal use: generic If CC is set to trunk use: native Generic/never 3600/4800 3600/7200 60/120 Value from 1-999 Value from 1-999 Yes/no Digits
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Time Condition for Ring Simultaneously
Time Condition for Skip Trunk Auth
Enable LDAP Enable T.38 UDPTL Max Contacts Enable WebRTC Alert-Info
Do Not Disturb DND Time Condition
Custom Auto answer Do Not Disturb Whitelist User Password First Name Last Name Email Address Language Phone Number Call-Barging Monitor Seamless Transfer Members
All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time Yes/no Yes/no Values from 1-10 Yes/no None/Ring 1/Ring2/Ring3/Ring 4/Ring 5/Ring 6/Ring 7/ Ring 8/Ring 9/Ring 10/bellcore-dr1/bellcore-dr2/ bellcore-dr3/ bellcore-dr4/ bellcore-dr5/custom Yes/no All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time Yes/no Empty/digits Alphanumeric characters. Alphanumeric without special characters. Alphanumeric without special characters. Email address Default/en/zh Digits Extensions allowed to call barging Extensions allowed to seamless transfer
Table 53: IAX extensions Imported File Example
Field
Supported values
Extension
Digits
Technology
IAX
Enable Voicemail
yes/no
CallerID Number
Digits
SIP/IAX Password
Alphanumeric characters
Voicemail Password
Digits
Skip Voicemail Password Verification
yes/no
Ring Timeout
Empty/ 3 to 600 (in second)
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SRTP Strategy Local Subnet 1 Local Subnet 2 Local Subnet 3 Local Subnet 4 Local Subnet 5 Local Subnet 6 Local Subnet 7 Local Subnet 8 Local Subnet 9 Local Subnet 10 Specific IP Address Skip Trunk Auth Codec Preference
Permission NAT Call Forward Busy Call Forward No Answer Call Forward Unconditional Require Call Token Max Number of Calls Dial Trunk Password Disable This Extension CFU Time Condition CFN Time Condition CFB Time Condition
Music On Hold Ring simultaneously External Number Time Condition for Ring Simultaneously
yes/no Allow All/Local Subnet Only/A Specific IP Address IP address/Mask IP address/Mask IP address/Mask IP address/Mask IP address/Mask IP address/Mask IP address/Mask IP address/Mask IP address/Mask IP address/Mask IP address yes/no/bytime PCMU,PCMA,GSM,G.726,G.722,G.729,H.264, H.265,ILBC,AAL2-G.72632,ADPCM,G.723,H.263,H.263p,vp8,opus Internal/Local/National/International yes/no Digits Digits Digits Yes/no/auto Values from 1-512 Digits Yes/no All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time Default/ringbacktone_default Yes/no Digits All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
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Time Condition for Skip Trunk Auth
Enable LDAP Limit Max time (s) Do Not Disturb DND Time Condition
Do Not Disturb Whitelist User Password First Name Last Name Email Address Language Phone Number Call-Barging Monitor Seamless Transfer Members
All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time Yes/no empty Yes/no All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time Empty/digits Alphanumeric characters. Alphanumeric without special characters. Alphanumeric without special characters. Email address Default/en/zh Digits Extensions allowed to call barging Extensions allowed to seamless transfer
Table 54: FXS Extensions Imported File Example
Field
Supported values
Extension
Digits
Technology
FXS
Analog Station
FXS1/FXS2
Enable Voicemail
yes/no
CallerID Number
Digits
Voicemail Password
Digits
Skip Voicemail Password Verification
yes/no
Ring Timeout
Empty/ 3 to 600 (in second)
Auto Record
yes/no
Fax Mode
None/Fax Gateway/Fax Detection
Skip Trunk Auth
Yes/no/bytime
Permission
Internal/Local/National/International
Call Forward Busy
Digits
Call Forward No Answer
Digits
Call Forward Unconditional
Digits
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Call Waiting Use # as SEND RX Gain TX Gain MIN RX Flash MAX RX Flash Enable Polarity Reversal Echo Cancellation 3-Way Calling Send CallerID After Dial Trunk Password Disable This Extension CFU Time Condition
CFN Time Condition
CFB Time Condition
Music On Hold CC Agent Policy
CC Monitor Policy CCBS Available Timer CCNR Available Timer CC Offer Timer CC Max Agents CC Max Monitors Ring simultaneously External Number Time Condition for Ring Simultaneously Time Condition for Skip Trunk Auth Enable LDAP Enable Hotline Hotline Type
Yes/no Yes/no Values from -306 Values from -306 Values from: 30 1000 Values from: 40 2000 Yes/no On/Off/32/64/128/256/512/1024 Yes/no 1/2 digits Yes/no All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time Default/ringbacktone_default If CC is disabled use: never If CC is set to normal use: generic If CC is set to trunk use: native Generic/never 3600/4800 3600/7200 60/120 Value from 1-999 Value from 1-999 Yes/no Digits All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time
Yes/no Yes/no Immediate hotline/delay hotline
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Hotline Number Limit Max time (s) Do Not Disturb DND Time Condition
Do Not Disturb Whitelist User Password First Name Last Name Email Address Language Phone Number Call-Barging Monitor Seamless Transfer Members
digits empty Yes/no All time/Office time/out of office time/holiday/out of holiday/out of office time or holiday/specific time Empty/digits Alphanumeric characters. Alphanumeric without special characters. Alphanumeric without special characters. Email address Default/en/zh Digits Extensions allowed to call barging Extensions allowed to seamless transfer
The CSV file should contain all the above fields, if one of them is missing or empty, the UCM630X will display the following error message for missing fields.
Figure 84: Import Error
Extension Details
Users can click on an extension number in the Extensions list page and quickly view information about it such as:
Extension: Shows the Extension number. Status: Shows the status of the extension. Presence status: Indicates the Presence Status of this extension. Terminal Type: Shows the Type of the terminal using this extension (SIP, FXS...etc.). Caller ID Name: Reveals the Caller ID Name configured on the extension. Messages: Shows the messages stats. IP and Port: The IP address and the ports of the device using the extension. Email status: Show the Email status (sent, to be sent...etc.). Ring Group: Indicates the ring groups that this extension belongs to. Call Queue: Indicates the Cal Queues that this extension belongs to. Call Queue (Dynamic): Indicates the Call Queues that this extension belongs to as a dynamic agent.
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Figure 85: Extension Details
E-mail Notification
Once the extensions are created with Email addresses, the PBX administrator can click on button "E-mail Notification" to send the account registration and configuration information to the user. Please make sure Email setting under Web GUISystem SettingsEmail Settings is properly configured and tested on the UCM630X before using "E-mail Notification".
When click on "More" > "E-mail Notification" button, the following message will be prompted in the web page. Click on OK to confirm sending the account information to all users' Email addresses.
Figure 86: E-mail Notification - Prompt Information
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The user will receive Email including account registration information as well as the GSWave Settings with the QR code:
Figure 87: Account Registration Information
Figure 88: GSWave Settings and QR Code
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Multiple Registrations per Extension
UCM630X supports multiple registrations per extension so that users can use the same extension on devices in different locations.
Figure 89: Multiple Registrations per Extension
This feature can be enabled by configuring option "Concurrent Registrations" under Web GUIExtension/TrunkEdit Extension. The default value is set to 1 for security purpose. Maximum is 10.
Figure 90: Extension - Concurrent Registration
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SMS Message Support
The UCM630X provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or GXV phones that supports SIP message, after an UCM630X account is registered on the end device, the user can send and receive SMS message. Please refer to the end device documentation on how to send and receive SMS message.
Figure 91: SMS Message Support
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EXTENSION GROUPS
The UCM630X extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the UCM630X. For example, when configuring "Enable Filter on Source Caller ID", users could select a group instead of each person's extension to assign. This feature simplifies the configuration process and helps manage and categorize the extensions for business environment.
Configure Extension Groups
Extension group can be configured via Web GUIExtension/TrunkExtension Groups.
Click on
to create a new extension group.
Click on
to edit the extension group.
Click on
to delete the extension group.
Select extensions from the list on the left side to the right side.
Figure 92: Edit Extension Group
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Click on
in order to change the ringing priority of the members selected on the group.
Using Extension Groups
Here is an example where the extension group can be used. Go to Web GUIExtension/TrunkOutbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and extension groups will show up for users to select.
Figure 93: Select Extension Group in Outbound Route
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ANALOG TRUNKS
Go to Web GUIExtension/TrunkAnalog Trunks to add and edit analog trunks.
Click on "Create New Analog Trunk" to add a new analog trunk. Click on to edit the analog trunk.
Click on to delete the analog trunk.
Analog Trunk Configuration
The analog trunk options are listed in the table below.
Table 55: Analog Trunk Configuration Parameters
FXO Port
Select the channel for the analog trunk. UCM6301: 1 channel UCM6302: 2 channels UCM6304: 4 channels UCM6308: 8 channels
Trunk Name
Specify a unique label to identify the trunk when listed in outbound rules, incoming rules and etc.
Advanced Options
SLA Mode
Enable this option to satisfy two primary use cases, which include emulating a simple key system and creating shared extensions on a PBX. Enable SLA Mode will disable polarity reversal.
Barge Allowed
The barge option specifies whether other stations can join a call in progress on this trunk. If enabled, the other stations can press the line button to join the call. The default setting is Yes.
Hold Access
The hold option specifies hold permissions for this trunk. If set to "Open", any station can place this trunk on hold and any other station is allowed to retrieve the call. If set to "Private", only the station that places the call on hold can retrieve the call. The default setting is Yes.
Enable Polarity Reversal
If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity reversal is used for signaling the disconnection of a phone line and the call will be considered as "Hangup" on a polarity reversal. The default setting is "No".
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Polarity on Answer Delay Current Disconnect Threshold (ms) Ring Timeout RX Gain TX Gain Use CallerID
Caller ID Scheme
Fax Mode
When FXO port answers the call, FXS may send a Polarity Reversal. If this interval is shorter than the value of "Polarity on Answer Delay", the Polarity Reversal will be ignored. Otherwise, the FXO will Onhook to disconnect the call. The default setting is 600ms.
This is the periodic time (in ms) that the UCM630X will use to check on a voltage drop in the line. The default setting is 200. The valid range is 50 to 3000.
Configure the ring timeout (in ms). Trunk (FXO) devices must have a timeout to determine if there was a Hangup before the line is answered. This value can be used to configure how long it takes before the UCM630X considers a non-ringing line with Hangup activity. The default setting is 8000. Configure the RX gain for the receiving channel of analog FXO port. The valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0. Configure the TX gain for the transmitting channel of analog FXO port. The valid range is from -13.5 (dB) to + 12.0 (dB). The default setting is 0. Configure to enable CallerID detection. The default setting is "Yes". Select the Caller ID scheme for this trunk.
Bellcore/Telcordia. ETSI-FSK During Ringing ETSI-FSK Prior to Ringing with DTAS ETSI-FSK Prior to Ringing with LR ETSI-FSK Prior to Ringing with RP ETSI-DTMF During Ringing ETSI-DTMF Prior to Ringing with DTAS ETSI-DTMF Prior to Ringing with LR ETSI-DTMF Prior to Ringing with RP SIN 227-BT NTT Japan Auto Detect If you are not sure which scheme to choose, please select "Auto Detect". The default setting is "Bellcore/Telcordia". Configures how faxes to this extension will be handled. None: Faxes will not be processed. Fax Gateway: Faxes to this extension will be processed and converted from
T.30 to T.38 or vice-versa. FXS/FXO ports only. The default setting is None.
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FXO Dial Delay (ms)
Configure the time interval between off-hook and first dialed digit for outbound calls.
Auto Record
Enable automatic recording for the calls using this trunk. The default setting is disabled. The recording files can be accessed under Web GUICDRRecording Files.
If selected, the trunk will be disabled and incoming/Outgoing calls via this trunk will Disable This Trunk
not be possible.
This is to implement analog trunk outbound line selection strategy. Three options are available:
DAHDI Out Line Selection
Ascend When the call goes out from this analog trunk, it will always try to use the first idle FXO port. The port order that the call will use to go out if UCM6302 is used would be port 1port 2. Every time it will start with port 1 (if it is idle).
Poll When the call goes out from this analog trunk, it will use the port that is not used last time. And it will always use the port in the order of port 121212..., following the last port being used in case UCM6302 is used.
Descend When the call goes out from this analog trunk, it will always try to use the last idle FXO port. The port order that the call will use to go out if UCM6302 is used would be port 2port 1. Every time it will start with port 2 (if it is idle).
The default setting is "Ascend" mode. The Non-Linear Processing (NLP) in echo cancellation helps to remove/suppress residual echo components that could not be removed by the LEC (Line Echo Canceller). Following modes are supported:
Default: The NLP limits the signal level to the background noise level when active, and the background noise level adjustment is low.
Echo Cancellation Mode
High Noise Level Adjustment: The NLP limits the signal level to the background noise level when active, and the background noise level adjustment is high.
Noise Masking: The NLP sends sign noise when active, and the background noise level adjustment is high.
White Noise Injection: The NLP injects white noise when active. The level corresponds to the background noise level at Sin, and the background noise level adjustment is high.
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Allows external numbers the option to get directed to the extension that last called them.
Direct Callback
Tone Settings Busy Detection Busy Tone Count Congestion Detection Congestion Count Tone Country
Busy Tone
Congestion Tone
For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option "Direct Callback" is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.
Busy Detection is used to detect far end Hangup or for detecting busy signal. The default setting is "Yes". If "Busy Detection" is enabled, users can specify the number of busy tones to be played before hanging up. The default setting is 2. Better results might be achieved if set to 4, 6 or even 8. Please note that the higher the number is, the more time is needed to Hangup the channel. However, this might lower the probability to get random Hangup. Congestion detection is used to detect far end congestion signal. The default setting is "Yes". If "Congestion Detection" is enabled, users can specify the number of congestion tones to wait for. The default setting is 2. Select the country for tone settings. If "Custom" is selected, users could manually configure the values for Busy Tone and Congestion Tone. The default setting is "United States of America (USA)". Syntax: f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]]; Frequencies are in Hz and cadence on and off are in ms. Frequencies Range: [0, 4000) Busy Level Range: (-300, 0) Cadence Range: [0, 16383]. Select Tone Country "Custom" to manually configure Busy Tone value. Default value: f1=480@-50,f2=620@-50,c=500/500 Syntax: f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]]; Frequencies are in Hz and cadence on and off are in ms. Frequencies Range: [0, 4000) Busy Level Range: (-300, 0) Cadence Range: [0, 16383].
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PSTN Detection
Select Tone Country "Custom" to manually configure Busy Tone value. Default value: f1=480@-50,f2=620@-50,c=250/250
Click on "Detect" to detect the busy tone, Polarity Reversal and Current Disconnect by PSTN. Before the detecting, please make sure there are more than one channel configured and working properly. If the detection has busy tone, the "Tone Country" option will be set as "Custom".
PSTN Detection
The UCM630X provides PSTN detection function to help users detect the busy tone, Polarity Reversal and Current Disconnect by making a call from the PSTN line to another destination. The detecting call will be answered and up for about 1 minute. Once done, the detecting result will show and can be used for the UCM630X settings.
1. Go to UCM630X Web GUIExtension/TrunkAnalog Trunks page. 2. Click to edit the analog trunk created for the FXO port. 3. In the window to edit the analog trunk, go to "Tone Settings" section and there are two methods to set the
busy tone.
Tone Country. The default setting is "United States of America (USA)". PSTN Detection.
Figure 94: UCM630X FXO Tone Settings
4. Click on "Detect" to start PSTN detection.
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Figure 95: UCM630X PSTN Detection
If there are two FXO ports connected to PSTN lines, use the following settings for auto-detection. Detect Model: Auto Detect. Source Channel: The source channel to be detected. Destination Channel: The channel to help detecting. For example, the second FXO port. Destination Number: The number to be dialed for detecting. This number must be the actual PSTN number for the FXO port used as the destination channel.
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Figure 96: UCM630X PSTN Detection: Auto Detect
If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.
Figure 97: UCM630X PSTN Detection: Semi-Auto Detect
Detect Model: Semi-auto Detect. Source Channel: The source channel to be detected. Destination Number: The number to be dialed for detecting. This number could be a cell phone number or other PSTN number that can be reached from the source channel PSTN number.
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5. Click "Detect" to start detecting. The source channel will initiate a call to the destination number. For "Auto Detect", the call will be automatically answered. For "Semi-auto Detect", the UCM630X Web GUI will display prompt to notify the user to answer or hang up the call to finish the detecting process.
6. Once done, the detected result will show. Users could save the detecting result as the current UCM630X settings.
Table 56: PSTN Detection for Analog Trunk
Select "Auto Detect" or "Semi-auto Detect" for PSTN detection.
Detect Model
Auto Detect Please make sure two or more channels are connected to the UCM630X and in idle status before starting the detection. During the detection, one channel will be used as caller (Source Channel) and another channel will be used as callee (Destination Channel). The UCM630X will control the call to be established and hang up between caller and callee to finish the detection.
Semi-auto Detect Semi-auto detection requires answering or hanging up the call manually. Please make sure one channel is connected to the UCM630X and in idle status before starting the detection. During the detection, source channel will be used as caller and send the call to the configured Destination Number. Users will then need follow the prompts in Web GUI to help finish the detection.
Source Channel Destination Channel Destination Number Dump Call Progress Tone File
The default setting is "Auto Detect". Select the channel to be detected. Select the channel to help detect when "Auto Detect" is used. Configure the number to be called to help the detection.
Choose whether to save the calling tone file, it is not checked by default.
---------------------------------------------------------------------------------------------------------------------------------------------------
Note: The PSTN detection process will keep the call up for about 1 minute. If "Semi-auto Detect' is used, please pick up the call only after informed from the Web GUI prompt. Once the detection is successful, the detected parameters "Busy Tone", "Polarity Reversal" and "Current
Disconnect by PSTN" will be filled into the corresponding fields in the analog trunk configuration. ---------------------------------------------------------------------------------------------------------------------------------------------------
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VOIP TRUNKS
VoIP Trunk Configuration
VoIP trunks can be configured in UCM630X under Web GUIExtension/TrunkVoIP Trunks. Once created, the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and Options to edit/detect the trunk.
Click on "Add SIP Trunk" or "Add IAX Trunk" to add a new VoIP trunk.
Click on
to configure detailed parameters for the VoIP trunk.
Click on Click on
to configure Direct Outward Dialing (DOD) for the SIP Trunk. to start LDAP Sync.
Click on
to delete the VoIP trunk.
For VoIP trunk example, please refer to the document in the following link: http://www.grandstream.com/sites/default/files/Resources/ucm6xxx_sip_trunk_guide.pdf
The VoIP trunk options are listed in the table below.
Type Provider Name Host Name Keep Original CID Keep Trunk CID
Table 57: Create New SIP Trunk
Select the VoIP trunk type. Peer SIP Trunk Register SIP Trunk Configure a unique label (up to 64 character) to identify this trunk when listed in outbound rules, inbound rules etc.
Configure the IP address or URL for the VoIP provider's server of the trunk.
Keep the CID from the inbound call when dialing out. This setting will override "Keep Trunk CID" option. Please make sure that the peer PBX at the other side supports to match user entry using "username" field from authentication line.
If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No".
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Turn on this setting when the PBX is using public IP and communicating with
NAT
devices behind NAT. If there is one-way audio issue, usually it is related to NAT
configuration or SIP/RTP port support on the firewall.
Disable This Trunk TEL URI
If checked, the trunk will be disabled. Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider.
If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the RequestLine and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.
Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored.
Caller ID Number
Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call:
CallerID Name
Need Registration Allow outgoing calls if registration fails Username
Password Auth ID
Auto Record
From user (Register Trunk Only) CID from inbound call (Keep Original CID Enabled) Trunk Username/CallerID (Keep Trunk CID Enabled) DOD Extension CallerID Number Trunk Username/CallerID (Keep Trunk CID Disabled) Global Outbound CID. Configure the new name of the caller when the extension has no CallerID Name configured. Select whether the trunk needs to register on the external server or not when "Register SIP Trunk" type is selected. The default setting is No. Uncheck to block outgoing calls if registration fails. If "Need Registration" option is unchecked, this setting will be ignored. Enter the username to register to the trunk from the provider when "Register SIP Trunk" type is selected. Enter the password to register to the trunk from the provider when "Register SIP Trunk" is selected. Enter the Authentication ID for "Register SIP Trunk" type. Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUICDRRecording Files.
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Direct Callback
RemoteConnect Mode
Allows external numbers the option to get directed to the extension that last called them. For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option "Direct Callback" is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.
If enabled, the RemoteConnect related parameters will be set synchronously. Please make sure the trunk host is allocated by GDMS or it supports TLS.
Basic Settings Provider Name Host Name
Transport
SIP URI Scheme When Using TLS Keep Original CID Keep Trunk CID NAT
Disable This Trunk
TEL URI
Table 58: SIP Register Trunk Configuration Parameters
Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc. Configure the IP address or URL for the VoIP provider's server of the trunk. Configure the SIP transport protocol to be used in this trunk. The default setting is "UDP". UDP TCP TLS When TLS is selected as Transport for register trunk, users can select between SIP and SIPS URI scheme Keep the CID from the inbound call when dialing out. This setting will override "Keep Trunk CID" option. Please make sure that the peer PBX at the other side supports to match user entry using "username" field from authentication line. If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No". Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port configuration on the firewall. If selected, the trunk will be disabled. Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider. If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the RequestLine and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.
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Need Registration Allow outgoing calls if registration failure CallerID Name From Domain
From User
Username
Select whether the trunk needs to register on the external server or not when "Register SIP Trunk" type is selected. The default setting is No.
If enabled outgoing calls even if the registration to this trunk fail will still be able to go through. Note that if we uncheck "Need Registration" option, this option will be ignored.
Configure the new name of the caller when the extension has no CallerID Name configured. Configure the actual domain name where the extension comes from. This can be used to override the "From" Header. For example, "trunk.UCM630X.provider.com" is the From Domain in From Header: sip:1234567@trunk.UCM630X.provider.com. Configure the actual username of the extension. This can be used to override the "From" Header. There are cases where there is a single ID for registration (single trunk) with multiple DIDs. For example, "1234567" is the From User in From Header: sip:1234567@trunk.UCM630X.provider.com. Enter the username to register to the trunk from the provider when "Register SIP Trunk" type is selected.
Password
Enter the password to register to the trunk when "Register SIP Trunk" is selected.
Auth ID Auth Trunk Auto Record RemoteConnect Mode
Direct Callback
Enter the Authentication ID for "Register SIP Trunk" type.
If enabled, the UCM will send 401 response to the incoming call to authenticate the trunk. Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUICDRRecording Files.
If enabled, the RemoteConnect related parameters will be set synchronously. Please make the trunk host is allocated by GDMS or it supports TLS.
Allows external numbers the option to get directed to the extension that last called them. For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option "Direct Callback" is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.
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Advanced Settings Codec Preference Send PPI Header PPI Mode Send PAI Header
PAI Header Send Anonymous DOD As From Name
Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.265, H.263, H.263p and VP8.
If enabled, the SIP INVITE message sent to the trunk will contain PPI (P-PreferredIdentity) header. The default setting is "No". Note: "Send PPI Header" and "Send PAI Header" cannot be enabled at the same time. Only one of the two headers can be contained in SIP INVITE message. Default Include the trunk's preferred CID (configured in Basic Settings) in the PPI Header. Original CID Include the original CID in the PPI Header. DOD Number Include the trunk's DOD number in the PPI Header. If no DOD number has been set, the trunk's preferred CID will be used.
If enabled, the SIP INVITE message sent to the trunk will contain PAI (P-AssertedIdentity) header including configured PAI Header. The default setting is "No". Note: "Send PPI Header" and "Send PAI Header" cannot be enabled at the same time. Only one of the two headers can be contained in the SIP INVITE message. If "Send PAI Header" is enabled and "PAI Header" is configured as "123456" for instance, the PAI header in the SIP message sent from the UCM will contain "123456". If "Send PAI Header" is enabled and "PAI Header" is configured as "empty", the PAI header in the SIP message sent from the UCM will contain the original CID. Note: "Send PAI Header" needs to be enabled to use this feature If checked, the "From" header in outgoing INVITE message will be set to anonymous. If enabled and "From User" is configured, the INVITE's From header will contain the DOD number.
Passthrough PAI Header
Send PANI Header
If checked and option "Send PAI Header" not checked, the PAI header will be passthrough from one side to the other side.
If checked, the INVITE and REGISTER sent to the trunk will contain P-AccessNetwork-Info header.
Access Network Info The access network information in the P-Access-Network-Info header.
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Send Anonymous
If checked, the "From" header in outgoing INVITE message will be set to anonymous.
Outbound Proxy Support Outbound Proxy
Remove OBP from Route
DID Mode GIN Registration
Select to enable outbound proxy in this trunk. The default setting is "No". When outbound proxy support is enabled, enter the IP address or URL of the outbound proxy. It is used to set if the phone system will remove outbound proxy URI from the route header. If is set to "Yes", it will remove the route header from SIP requests. The default setting is "No". Configure where to get the destination ID of an incoming SIP call, from SIP Request-line or To-header. The default is set to "Request-line". If enabled, the UCM will send a GIN REGISTER (generate implicit numbers). Configure the default DTMF mode when sending DTMF on this trunk.
Default: The global setting of DTMF mode will be used. The global setting for DTMF Mode setting is under Web GUIPBX SettingsSIP SettingsToS.
DTMF Mode
RFC4733: Send DTMF using RFC4733. Info: Send DTMF using SIP INFO message.
Inband: Send DTMF using inband audio. This requires 64-bit codec, i.e., PCMU and PCMA.
Enable Heartbeat Detection
Heartbeat Frequency
The Maximum Number of Call Lines Packet Loss Retransmission Audio FEC Video FEC
ICE support
Auto: Send DTMF using RFC4733 if offered. Otherwise, inband will be used.
If enabled, the UCM630X will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No". When "Enable Heartbeat Detection" option is set to "Yes", configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds. The maximum number of concurrent calls using the trunk. The default settings 0, which means no limit. Configure to enable Packet Loss Retransmission.
Configure to enable Forward Error Correction (FEC) for audio. Configure to enable Forward Error Correction (FEC) for video. Toggles ICE support. For peer trunks, ICE support will need to be enabled on the other end.
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FECC SRTP Enable T.38 UDPTL STIR/SHAKEN CC Settings Enable CC
CC Max Agents
CC Max Monitors
Configure to enable Far-end Camera Control Enable SRTP for the VoIP trunk. The default setting is "No". Enable or disable T.38 UDPTL support. Block disturbance calls, this function needs to be supported by the opposite end.
If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason.
Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel is allowed to make. The minimum value is 1. Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.
Basic Settings Provider Name Host Name Auto Record
Keep Original CID Keep Trunk CID NAT
Disable This Trunk TEL URI
Table 59: SIP Peer Trunk Configuration Parameters
Configure a unique label to identify this trunk when listed in outbound rules, inbound rules and etc. Configure the IP address or URL for the VoIP provider's server of the trunk. Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed under Web GUICDRRecording Files. Keep the CID from the inbound call when dialing out, this setting will override "Keep Trunk CID" option. Please make sure that the peer PBX at the other side supports to match user entry using "username" field from authentication line. If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No". Turn on this option when the PBX is using public IP and communicating with devices behind NAT. If there is one-way audio issue, usually it is related to NAT configuration or SIP/RTP port configuration on the firewall. If selected, the trunk will be disabled. Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message (REGISTER message with expires=0) to the SIP provider. If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-
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Caller ID Number
CallerID Name
Transport RemoteConnect Mode
Direct Callback
Advanced Settings Codec Preference Send PPI Header Send PAI Header Passthrough PAI Header
Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.
Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored. Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call : CID from inbound call (Keep Original CID Enabled) Trunk
Username/CallerID (Keep Trunk CID Enabled) DOD Extension CallerID Number Trunk Username/CallerID (Keep Trunk CID Disabled) Global Outbound CID. Configure the name of the caller to be displayed when the extension has no CallerID Name configured. Configure the SIP transport protocol to be used in this trunk. The default setting is "UDP". UDP TCP TLS If enabled, the RemoteConnect related parameters will be set synchronously. Please make the trunk host is allocated by GDMS or it supports TLS. Allows external numbers the option to get directed to the extension that last called them. For Example: User 2002 has dialed external number 061234575 but they were not reachable, once they have received missed call notification on their phone, they would mostly call back the number, if the option "Direct Callback" is enabled then they will be directly bridged to user 2002 regardless of the configured inbound destination.
Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.265, H.263, H.263p and VP8. If checked, the invite message sent to trunks will contain PPI (P-Preferred-Identity) Header. If checked, the INVITE, 18x and 200 SIP messages sent to trunks will contain PAsserted-Identity (PAI) header. It is not possible to send both PPI and PAI headers. If enabled and "Send PAI Header" is disabled, PAI headers will be preserved as calls pass through the UCM.
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Send PANI Header Send Anonymous DID Mode
If checked, the INVITE sent to the trunk will contain P-Access-Network-Info header. If checked, the "From" header in outgoing INVITE message will be set to anonymous. Configure where to get the destination ID of an incoming SIP call, from SIP Request-line or To-header. The default is set to "Request-line". Configure the default DTMF mode when sending DTMF on this trunk.
Default: The global setting of DTMF mode will be used. The global setting for DTMF Mode setting is under Web GUIPBX SettingsSIP SettingsToS.
DTMF Mode
RFC4733: Send DTMF using RFC4733. Info: Send DTMF using SIP INFO message.
Inband: Send DTMF using inband audio. This requires 64-bit codec, i.e., PCMU and PCMA.
Enable Heartbeat Detection
Heartbeat Frequency
Maximum Number of Call Lines Packet Loss Retransmission Audio FEC Video FEC ICE support FECC SRTP IPVT Mode
Sync LDAP Enable
Auto: Send DTMF using RFC4733 if offered. Otherwise, inband is used. If enabled, the UCM630X will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No". When "Enable Heartbeat Detection" option is set to "Yes", configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds. The maximum number of concurrent calls using the trunk. The default settings 0, which means no limit. Configure to enable Packet Loss Retransmission.
Configure to enable Forward Error Correction (FEC) for audio. Configure to enable Forward Error Correction (FEC) for video. Toggles ICE support. For peer trunks, ICE support will need to be enabled on the other end. Configure to enable Far-end Camera Control Enable SRTP for the VoIP trunk. The default setting is "No". Configures the UCM to be used exclusively for IPVT. Warning: This will lock out certain UCM features. Automatically sync local LDAP phonebooks to a remote peer (SIP peer trunk only). To ensure successful syncing, the remote peer must also enable this service and set the same password as the local UCM. Port 873 is used by default.
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Sync LDAP Password
Password used for LDAP phonebook encryption and decryption. The password must be the same for both peers to ensure successful syncing.
Specify an outbound rule for LDAP sync feature. The UCM630X will automatically LDAP Outbound Rule
modify the remote contacts by adding prefix parsed from this rule.
LDAP Dialed Prefix
Specify the prefix for LDAP sync feature. The UCM630X will automatically modify the remote contacts by adding this prefix.
LDAP Last Sync Date The last successful sync date.
Enable T.38 UDPTL Enable or disable T.38 UDPTL support.
STIR/SHAKEN
Block disturbance calls, this function needs to be supported by the opposite end.
CC Settings
Enable CC
If enabled, the system will automatically alert the user when a called party is available, given that a previous call to that party failed for some reason.
CC Max Agents
Configure the maximum number of CCSS agents which may be allocated for this channel. In other words, this number serves as the maximum number of CC requests this channel can make. The minimum value is 1.
CC Max Monitors
Configure the maximum number of monitor structures which may be created for this device. In other words, this number tells how many callers may request CC services for a specific device at one time. The minimum value is 1.
Type
Provider Name Host Name Keep Trunk CID Username Password Disable This Trunk Caller ID Number
Table 60: Create New IAX Trunk
Select the VoIP trunk type. Peer IAX Trunk Register IAX Trunk Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc. Configure the IP address or URL for the VoIP provider's server of the trunk. If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No". Enter the username to register to the trunk from the provider when "Register IAX Trunk" type is selected. Enter the password to register to the trunk from the provider when "Register IAX Trunk" type is selected. If selected, the trunk will be disabled. Number that the trunk will try to use when making outbound calls. CID priority from highest to lowest is as follows: From User (register trunk only) >>> Inbound Call CID (if Keep Original CID is
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CallerID Name
enabled and the call is originally from another trunk) >>> Trunk CID (Keep Trunk CID enabled) >>> DOD CID >>> Extension CID >>> Register Trunk Username (Keep Trunk CID disabled) >>> Global Outbound CID. Note 1: Certain providers may ignore this CID. Note 2: If this CID contains asterisk (*), call recordings from this trunk might be lost when saving them to NAS storage.
Configure the new name of the caller when the extension has no CallerID Name configured.
Basic Settings Provider Name Host Name Keep Trunk CID Disable This Trunk
Caller ID
CallerID Name Username Password Advanced Settings Codec Preference
Table 61: IAX Register Trunk Configuration Parameters
Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc. Configure the IP address or URL for the VoIP provider's server of the trunk. If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No". If selected, the trunk will be disabled. Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored. Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call : From user (Register Trunk Only) CID from inbound call (Keep Original CID Enabled) Trunk Username/CallerID (Keep Trunk CID Enabled) DOD Extension CallerID Number Trunk Username/CallerID (Keep Trunk CID Disabled) Global Outbound CID. Configure the name of the caller to be displayed when the extension has no CallerID Name configured. Enter the username to register to the trunk from the provider. Enter the password to register to the trunk from the provider.
Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.265, H.263, H.263p and VP8.
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Enable Heartbeat Detection
If enabled, the UCM630X will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No".
Heartbeat Frequency
When "Enable Heartbeat Detection" option is set to "Yes", configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds.
Maximum Number of The maximum number of concurrent calls using the trunk. The default settings 0,
Call Lines
which means no limited.
Basic Settings Provider Name Host Name Keep Trunk CID Disable This Trunk
Table 62: IAX Peer Trunk Configuration Parameters
Configure a unique label to identify this trunk when listed in outbound rules, inbound rules etc. Configure the IP address or URL for the VoIP provider's server of the trunk. If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No". If selected, the trunk will be disabled. Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored.
Caller ID
Important Note: When making outgoing calls, the following priority order rule will be used to determine which CallerID will be set before sending out the call :
CallerID Name Advanced Settings
Codec Preference
Enable Heartbeat Detection
Heartbeat Frequency
CID from inbound call (Keep Original CID Enabled) Trunk Username/CallerID (Keep Trunk CID Enabled) DOD Extension CallerID Number Trunk Username/CallerID (Keep Trunk CID Disabled) Global Outbound CID.. Configure the name of the caller to be displayed when the extension has no CallerID Name configured.
Select audio and video codec for the VoIP trunk. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM, H.264, H.265, H.263, H.263p and VP8. If enabled, the UCM630X will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No". When "Enable Heartbeat Detection" option is set to "Yes", configure the interval (in seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds.
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Maximum Number of The maximum number of concurrent calls using the trunk. The default settings 0,
Call Lines
which means no limited.
Trunk Groups
Users can create VoIP Trunk Groups to register and easily apply the same settings on multiple accounts within the same SIP server. This can drastically reduce the amount of time needed to manage accounts for the same server and improve the overall cleanliness of the web UI.
Figure 98: Trunk Group
Once creating the new trunk group and configuring the SIP settings, users can add multiple accounts within the
configured SIP server by pressing fields.
button and configuring the username, password, and authentication ID
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Figure 99: Trunk Group Configuration
Direct Outward Dialing (DOD)
The UCM630X provides Direct Outward Dialing (DOD) which is a service of a local phone company (or local exchange carrier) that allows subscribers within a company's PBX system to connect to outside lines directly.
Example of how DOD is used: Company ABC has a SIP trunk. This SIP trunk has 4 DIDs associated to it. The main number of the office is routed to an auto attendant. The other three numbers are direct lines to specific users of the company. Now when a user makes an outbound call their caller ID shows up as the main office number. This poses a problem as the CEO would like their calls to come from their direct line. This can be accomplished by configuring DOD for the CEO's extension. Steps to configure DOD on the UCM630X:
1. To setup DOD go to UCM630X Web GUIExtension/TrunkVoIP Trunks page.
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2. Click
to access the DOD options for the selected SIP Trunk.
3. Click "Add DOD" to begin your DOD setup
4. For "DOD Number" enter one of the numbers (DIDs) from your SIP trunk provider. In the example above Company ABC received 4 DIDs from their provider. ABC will enter in the number for the CEO's direct line.
5. Set the DOD name and If extension number need to be appended to the DID number click on "Add Extension".
6. Select an extension from the "Available Extensions" list. Users have the option of selecting more than one extension. In this case, Company ABC would select the CEO's extension. After making the selection, click on the button to move the extension(s) to the "Selected Extensions" list.
7. Click "Save" at the bottom.
Figure 100: DOD extension selection
Once completed, the user will return to the EDIT DOD page that shows all the extensions that are associated to a particular DOD.
Figure 101: Edit DOD
Note: Users can import and export DOD files.
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SLA STATION
The UCM630X supports SLA that allows mapping the key with LED on a multi-line phone to different external lines. When there is an incoming call and the phone starts to ring, the LED on the key will flash in red and the call can be picked up by pressing this key. This allows users to know if the line is occupied or not. The SLA function on the UCM630X is like BLF but SLA is used to monitor external line i.e., analog trunk on the UCM630X. Users could configure the phone with BLF mode on the MPK to monitor the analog trunk status or press the line key pick up call from the analog trunk on the UCM630X.
Create/Edit SLA Station
SLA Station can be configured on Web GUIExtension/TrunkSLA Station.
Click on
Figure 102: SLA Station
to add an SLA Station.
Click on
to edit the SLA Station. The following table shows the SLA Station configuration parameters.
Click on
to delete the SLA Station.
Table 63: SLA Station Configuration Parameters
Station Name
Configure a name to identify the SLA Station.
Station
Specify a SIP extension as a station that will be using SLA.
Available SLA Trunks Existing Analog Trunks with SLA Mode enabled will be listed here.
Associated SLA Trunks
Select a trunk for this SLA from the Available SLA Trunks list. Click on to arrange the order. If there are multiple trunks selected, when there are
calls on those trunks at the same time, pressing the LINE key on the phone will pick up the call on the first trunk here.
SLA Station Options
Ring Timeout
Configure the time (in seconds) to ring the station before the call is considered unanswered. No timeout is set by default. If set to 0, there will be no timeout.
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Ring Delay Hold Access
Configure the time (in seconds) for delay before ringing the station when a call first coming in on the shared line. No delay is set by default. If set to 0, there will be no delay.
This option defines the competence of the hold action for one particular trunk. If set to "open", any station could hold a call on that trunk or resume one held session; if set to "private", only the station that places the trunk call on hold could resume the session. The default setting is "open".
Sample Configuration
1. On the UCM630X, go to Web GUIExtension/TrunkAnalog Trunks page. Create analog trunk or edit the existing analog trunk. Make sure "SLA Mode" is enabled for the analog trunk. Once enabled, this analog trunk will be only available for the SLA stations created under Web GUIExtension/TrunkSLA Station page.
Figure 103: Enable SLA Mode for Analog Trunk
2. Click on "Save". The analog trunk will be listed with trunk mode "SLA".
Figure 104: Analog Trunk with SLA Mode Enabled
3. On the UCM630X, go to Web GUIExtension/TrunkSLA Station page, click on "Add". Please refer to section [Create/Edit SLA Station] for the configuration parameters. Users can create one or more SLA stations to monitor the analog trunk. The following figure shows two stations, 1002 and 1005, are configured to be associated with SLA trunk "fxo1".
Figure 105: SLA Example - SLA Station
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4. On the SIP phone 1, configure to register UCM630X extension 1002. Configure the MPK as BLF mode and the value must be set to "extension_trunkname", which is 1002_fxo1 in this case.
5. On the SIP phone 2, configure to register UCM630X extension 1005. Configure the MPK as BLF mode and value must be set to "extension_trunkname", which is 1005_fxo1 in this case.
Figure 106: SLA Example - MPK Configuration
Now the SLA station is ready to use. The following functions can be achieved by this configuration.
Making an outbound call from the station/extension, using LINE key When the extension is in idle state, pressing the line key for this extension on the phone to off hook. Then dial the station's extension number, for example, dial 1002 on phone 1 (or dial 1005 on phone 2), to hear the dial tone. Then the users could dial external number for the outbound call.
Making an outbound call from the station/extension, using BLF key When the extension is in idle state, pressing the MPK and users could dial external numbers directly.
Answering call using LINE key When the station is ringing, pressing the LINE key to answer the incoming call.
Barging-in active call using BLF key When there is an active call between an SLA station and an external number using the SLA trunk, other SLA stations monitoring the same trunk could join the call by pressing the BLF key if "Barge Allowed" is enabled for the analog trunk.
Hold/UnHold using BLF key If the external line is previously put on hold by an SLA station, another station that monitors the same SLA trunk could UnHold the call by pressing the BLF key if "Hold Access" is set to "open" on the analog trunk and the SLA station.
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CALL ROUTES
Outbound Routes
In the following sections, we will discuss the steps and parameters used to configure and manage outbound rules in UCM630X, these rules are the regulating points for all external outgoing calls initiated by the UCM through all types of trunks: SIP, Analog and Digital.
Configuring Outbound Routes
In the UCM630X, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern. This allows different patterns to be dialed through different trunks (e.g., "Local" 7-digit dials through an FXO while "Long distance" 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to be used when the primary trunk fails.
Go to Web GUIExtension/TrunkOutbound Routes to add and edit outbound rules.
Click on Click on Click on
to add a new outbound route. to edit the outbound route. to delete the outbound route.
On the UCM630X, the outbound route priority is based on "Best matching pattern". For example, the UCM630X has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured. When dialing 1000 for outbound call, outbound route B will always be used first. This is because pattern 10xx is a better match than pattern 1xxx. Only when there are multiple outbound routes with the same pattern configured.
Table 64: Outbound Route Configuration Parameters
Configure the name of the calling rule (e.g., local, long_distance, and etc.). Letters, Outbound Rule Name
digits, _ and - are allowed.
All patterns are prefixed by "_" character, but please do not enter more than one "_" at the beginning. All patterns can add comments, such as "_pattern /* comment */". In patterns, some characters have special meanings:
Pattern
[12345-9] ... Any digit in the brackets. In this example, 1,2,3,4,5,6,7,8,9 is allowed.
N ... Any digit from 2-9. . ... Wildcard, matching one or more characters. ! ... Wildcard, matching zero or more characters immediately. X ... Any digit from 0-9. Z ... Any digit from 1-9. - ... Hyphen is to connect characters and it will be ignored.
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Disable This Route Password Local Country Code Call Duration Limit Maximum Call Duration Warning Time Auto Record Warning Repeat Interval PIN Groups PIN Groups with Privilege Level
Privilege Level
Enable Filter on Source Caller ID
[] Contain special characters ([x], [n], [z]) represent letters x, n, z. After creating the outbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore. However, the route settings will remain in UCM. Users can enable it again when it is needed. Configure the password for users to use this rule when making outbound calls. If your local country code is affected by the outbound blacklist, please enter it here to bypass the blacklist. Enable to configure the maximum duration for the call using this outbound route. Configure the maximum duration of the call (in seconds). The default setting is 0, which means no limit. Configure the warning time for the call using this outbound route. If set to x seconds, the warning tone will be played to the caller when x seconds are left to end the call. If enabled, calls using this route will automatically be recorded. Configure the warning repeat interval for the call using this outbound route. If set to X seconds, the warning tone will be played every x seconds after the first warning. Select a PIN Group If enabled and PIN Groups are used, Privilege Levels and Filter on Source Caller ID will also be applied.
Select privilege level for the outbound rule.
Internal: The lowest level required. All users can use this rule. Local: Users with Local, National, or International level can use this rule. National: Users with National or International level can use this rule. International: The highest level required. Only users with international level
can use this rule. Disable: The default setting is "Disable". If selected, only the matched source
caller ID will be allowed to use this outbound route.
Please be aware of the potential security risks when using "Internal" level, which means all users can use this outbound rule to dial out from the trunk.
When enabled, users could specify extensions allowed to use this outbound route. "Privilege Level" is automatically disabled if using "Enable Filter on Source Caller ID".
The following two methods can be used at the same time to define the extensions as the source caller ID.
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1. Select available extensions/extension groups from the list. This allows users to specify arbitrary single extensions available in the PBX.
2. Custom Dynamic Route: define the pattern for the source caller ID. This allows users to define extension range instead of selecting them one by one.
All patterns are prefixed with the "_". Special characters:
X: Any Digit from 0-9. Z: Any Digit from 1-9. N: Any Digit from 2-9. ".": Wildcard. Match one or more characters. "!": Wildcard. Match zero or more characters immediately. Example: [12345-9] - Any digit from 1 to 9. Note: Multiple patterns can be used. Patterns should be separated by comma ",". Example: _X. , _NNXXNXXXXX , _818X.
Attempt to use the configured outbound route CID. This CID will not be used if Outbound Route CID
DOD is configured.
Send This Call Through Trunk
Trunk
Select the trunk for this outbound rule.
Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk.
Strip Prepend Use Failover Trunk
Failover Trunk
Example: The users will dial 9 as the first digit of a long-distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed. Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.
Failover trunks can be used to make sure that a call goes through an alternate route when the primary trunk is busy or down. If "Use Failover Trunk" is enabled and "Failover trunk" is defined, the calls that cannot be placed via the regular trunk may have a secondary trunk to go through. UCM630X support up to 10 failover trunks.
Example: The user's primary trunk is a VoIP trunk and the user would like to use the PSTN when the VoIP trunk is not available. The PSTN trunk can be configured as the failover trunk of the VoIP trunk.
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Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk.
Strip
Example: The users will dial 9 as the first digit of a long-distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed.
Prepend
Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.
Time Condition
Time Condition Mode
Use Main Trunk or Failover Trunk: Use the Main Trunk and its settings during the configured time conditions. If the main trunk is unavailable, the Failover Trunk and its settings will be used instead. Use Specific Trunks: Use specific trunks during the configured time conditions. The Strip and Prepend settings of the Main Trunk will be used. If a trunk is unavailable during its time condition, no failover trunks will be used.
Outbound Blacklist
The UCM630X allows users to configure blacklist for outbound routes. If the dialing number matches the blacklist numbers or patterns, the outbound call will not be allowed. The outbound blacklist can be configured under UCM Web GUIExtension/TrunkOutbound Routes: Outbound Blacklist.
Users can configure numbers, patterns or select country code to add in the blacklist. Please note that the blacklist settings apply to all outbound routes.
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Figure 107: Country Codes
Note: Users can export outbound route blacklists and delete all blacklist entries. Additionally, users can also import blacklists for outbound routes.
Figure 108: Blacklist Import/Export
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Scheduled Sync
The UCM630X allows users to synchronize the outbound routes, this feature can be found on the WebGUIExtension/TrunkOutbound Routes Scheduled Sync.
Scheduled Sync Server Address File Name Sync Time Sync Frequency
Table 65: Outbound Routes/Scheduled Sync
Enable the Scheduled Sync feature Enter the TFTP server address. For example, "192.168.1.2:69". Specify the file name Enter the sync time (24hr format). Valid range is 0-23. Create new sync every x day(s). The valid range is 1 to 30.
PIN Groups
The UCM630X supports pin group. Once this feature is configured, users can apply pin group to specific outbound routes. When placing a call on pin protected outbound routes, caller will be asked to input the group pin number, this feature can be found on the WebGUIExtension/TrunkOutbound RoutesPIN Groups.
Name Record In CDR PIN Number PIN Name
Table 66: Outbound Routes/PIN Group
Specify the name of the group Specify whether to enable/disable record in CDR Specify the code that will asked once dialing via a trunk Specify the name of the PIN
Once user click on
the following figure shows to configure the new PIN.
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Figure 109: Create New PIN Group
The following screenshot shows an example of created PIN Groups and members:
Figure 110: PIN Members
Note:
If PIN group is enabled on outbound route level, password, privilege level and enable filter on source caller ID will be disabled, unless if you check the option "PIN Groups with Privilege Level" where you can use the PIN Groups and Privilege Level or PIN Groups and Enable Filter on Source Caller ID.
Figure 111: Outbound PIN
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If PIN group CDR is enabled, the call with PIN group information will be displayed as part of CDR under Account Code field.
Figure 112: CDR Record
- Importing PIN Groups from CSV files: User can also import PIN Groups by uploading CSV files for each group. To do this:
1. Navigate to Extension/TrunkOutbound RoutesPIN Groups and click on the "Upload" button.
Figure 113: Importing PIN Groups from CSV files
2. Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in error messages such as the one below:
Figure 114: Incorrect CSV File
3. To ensure a successful import, please follow the format in the sample image below
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Figure 115: CSV File Format
The top-left value (A1) is the PIN Group name. In this case, it is "ALPHA". Row 2 contains the labels for the modifiable fields: pin and pin_name. These values should not
be changed and will cause an upload error otherwise.
Rows 3+ contain the user-defined values with Column A holding the PINs and Column B holding the PIN names. PIN values must consist of at least four digits.
Once the file is successfully uploaded, the entry will be added to the list of PIN Groups.
Figure 116: CSV File Successful Upload
Inbound Routes
Inbound routes can be configured via Web GUIExtension/TrunkInbound Routes.
Click on
to add a new inbound route.
Click on "Blacklist" to configure blacklist for all inbound routes.
Click on
to edit the inbound route.
Click on to delete the inbound route.
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Inbound Rule Configurations
Trunks Inbound Route Name
Pattern
Table 67: Inbound Rule Configuration Parameters
Select the trunk to configure the inbound rule.
Configure the name of the Inbound Route. For example, "Local", "LongDistance" and etc.
All patterns are prefixed with the "_". Special characters:
X: Any Digit from 0-9. Z: Any Digit from 1-9. N: Any Digit from 2-9. ".": Wildcard. Match one or more characters. "!": Wildcard. Match zero or more characters immediately. Example: [12345-9] - Any digit from 1 to 9. The pattern can be composed of two parts, Pattern and CallerID Pattern. The first part is used to specify the dialed number while the second part is used to specify the caller ID and it is optional, if set it means only the extension with the specific caller ID can call in or call out. For example, pattern '_2XXX/1234' means the only extension with the caller ID '1234' can use this rule.
Note:
Multiple patterns can be used. Each pattern should be entered in new
line.
Users can add comments to the end of patterns to better organize and
keep track of complex rules by typing "/*" and "*/" before and after
each comment respectively
Example:
Pattern
CallerID Pattern
_X.
1000
_ NNXXNXXXXX /* 10-digit long distance */
1001
Disable This Route Seamless Transfer Whitelist Ringback tone
After creating the inbound route, users can choose to enable and disable it. If the route is disabled, it will not take effect anymore. However, the route settings will remain in UCM. Users can enable it again when it is needed.
Allows the selected extension to use this function. If an extension is busy, and a mobile phone is bound to that extension, the mobile phone can pick up calls to that extension. Choose the custom ring back tone to play when caller reaches the route.
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Auto Record Block Collect Call Alert-Info Fax Detection
Fax Destination
Prepend Trunk Name Set Caller ID Info
CalleID Number
If enabled, calls using this route will automatically be recorded. If enabled, collect calls will be blocked. Note: Collect calls are indicated by the header "P-Asserted-Service-Info: service-code=Backward Collect Call, P-Asserted-Service-Info: servicecode=Collect Call". Configure the Alert-Info, when UCM receives an INVITE request, the AlertInfo header field specifies an alternative ring tone to the UAS.
If enabled, fax signals from the trunk during a call will be detected.
Configures the destination of faxes. Extension: send the fax to the designated FXS/SIP extension (fax machine) or a FAX extension. Fax to Email: send the fax as an email attachment to the designated extension's email address. If the selected extension does not have an associated email address, it will be sent to the default email address configured in the Call Features->Fax/T.38->Fax Settings page.
Note: please make sure the sending email address is correctly configured in System Settings->Email Settings. If enabled, the trunk name will be added to the caller id name as the displayed caller id name. Manipulates Caller ID (CID) name and/or number within the call flow to help identify who is calling. When enabled two field will show allowing to manipulate the CalleID Number and the Caller ID Name. Configure the pattern-matching format to manipulate the numbers of incoming callers or to set a fixed CallerID number for calls that go through this inbound route.
${CALLERID(num)}: Default value which indicates the number of an incoming caller (CID). The CID will not be modified.
${CALLERID(num):n}: Skips the first n characters of a CID number, where n is a number.
${CALLERID(num):-n}: Takes the last n characters of a CID number, where n is a number.
${CALLERID(num):s:n}: Takes n characters of a CID number starting from s+1, where n is a number and s is a character position (e.g. ${CALLERID(num):2:7} takes 7 characters after the second character of a CID number).
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n${CALLERID(num)}: Prepends n to a CID number, where n is a number.
Default string is ${CALLERID(name)}which means the name of an incoming caller, it is a pattern-matching syntax format.
CallerID Name Enable Route-Level Inbound Mode
Inbound Mode
Inbound Mode Suffix Inbound Multiple Mode Dial Trunk
A${CALLERID(name)}B means Prepend a character 'A' and suffix a character 'B' to ${CALLERID(name)}.
Not using pattern-matching syntax means setting fix name to incoming caller.
Gives uses the ability to configure inbound mode per individual route. When enabled two field will show allowing to set the Inbound mode and the Inbound mode Suffix. Note: Global inbound mode must be enabled before users can configure route-level inbound mode Choose the inbound mode for this route. Note: Toggling the global inbound mode will not affect routes that have Routelevel Inbound Mode enabled. If all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change. Dial "Global Inbound Mode feature code + Inbound Mode Suffix" or a route's assigned suffix to toggle the route's inbound mode. The BLF subscribed to the inbound mode suffix can monitor the current inbound mode. Multiple mode allows user to switch between destinations of the inbound rule by feature codes. Configure related feature codes as described in [Inbound Route: Multiple Mode]. If this option is enabled, user can use feature code to switch between different modes/destinations. This option shows up only when "By DID" is selected. If enabled, the external users dialing in to the trunk via this inbound route can dial outbound call using the UCM's trunk. This option shows up only when "By DID" is selected.
Disable: Only the selected Extensions or Extension Groups are allowed to use this rule, when enabled Filter on Source Caller ID.
Privilege Level
Internal: The lowest level required. All users are allowed to use this rule, check this level might be risky for security purpose.
Local: User with Local level, National or International level are allowed to use this rule.
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Allowed DID Destination Default Destination Strip
National: Users with National or International Level are allowed to use this rule.
International: The highest level required. Only users with international level are allowed to use this rule.
This option shows up only when "By DID" is selected. This controls the destination that can be reached by the external caller via the inbound route. The DID destination are: Extension Conference Call Queue Ring Group Paging/Intercom Groups IVR Voicemail Groups Dial By Name All
Select the default destination for the inbound call. Extension Voicemail Conference Room Call Queue Ring Group Paging/Intercom Voicemail Group DISA IVR External Number By DID
When "By DID" is used, the UCM will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR and voicemail groups as configured in "DID destination". If the dialed number matches the DID pattern, the call will be allowed to go through. Dial By Name Callback Specify the number of digits to strip from the beginning of the DID. This is used when "By DID" is selected in "Default Destination".
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Prepend Time Condition Start Time End Time Date Week
Destination
Specify the digits to be prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped.
Select the start time "hour:minute" for the trunk to use the inbound rule. Select the end time "hour:minute" for the trunk to use the inbound rule. Select "By Week" or "By Day" and specify the date for the trunk to use the inbound rule. Select the day in the week to use the inbound rule. Select the destination for the inbound call under the defined time condition. Extension Voicemail Conference Room Call Queue Ring Group Paging/Intercom Voicemail Group DISA IVR By DID
When "By DID" is used, the UCM will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR and voicemail groups as configured in "DID destination". If the dialed number matches the DID pattern, the call will be allowed to go through. Configure the number of digits to be stripped in "Strip" option. Dial By Name External Number Callback
Inbound Route: Prepend Example
UCM630X now allows user to prepend digits to an inbound DID pattern, with strip taking precedence over prepend. With the ability to prepend digits in inbound route DID pattern, user no longer needs to create multiple routes for the same trunk to route calls to different extensions. The following example demonstrates the process:
1. If Trunk provides a DID pattern of 18005251163. 2. If Strip is set to 8, UCM630X will strip the first 8 digits.
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3. If Prepend is set to 2, UCM630X will then prepend a 2 to the stripped number, now the number become 2163.
4. UCM630X will now forward the incoming call to extension 2163.
Figure 117: Inbound Route feature: Prepend
Inbound Route: Multiple Mode In the UCM630X, the user can configure inbound route to enable multiple mode to switch between different destinations. The inbound multiple mode can be enabled under Inbound Route settings.
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Figure 118: Inbound Route - Multiple Mode
When Multiple Mode is enabled for the inbound route, the user can configure a "Default Destination" and a "Mode 1" destination for all routes. By default, the call coming into the inbound routes will be routed to the default destination.
SIP end devices that have registered on the UCM630X can dial feature code *62 to switch to inbound route "Mode 1" and dial feature code *61 to switch back to "Default Destination". Switching between different mode can be easily done without Web GUI login.
For example, the customer service hotline destination has to be set to a different IVR after 7PM. The user can dial *62 to switch to "Mode 1" with that IVR set as the destination before off work.
To customize feature codes for "Default Mode" and "Mode 1", click on
under
"Inbound Routes" page, check "Enable Inbound Multiple Mode" option and change "Inbound Default Mode" and "Inbound Mode 1" values (By default, *61 and *62 respectively).
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Figure 119: Inbound Route - Multiple Mode Feature Codes
Inbound Route: Route-Level Mode In the UCM630X, users can enable Route-Level Inbound Mode to switch between different destinations for each individual inbound route. The inbound Route-Level mode can be enabled under Inbound Route settings.
Figure 120: Inbound Route - Route-Level Mode
Global inbound mode must be enabled before configuring Route-Level Inbound Mode. Additionally, the Mode 1 must be configured as well.
When Route-Level Inbound Mode is enabled, the user can configure a "Default Destination" and a "Mode 1" destination for each specific route. By default, the call coming into this specific inbound route will be routed to the default destination.
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Users can toggle the route's inbound mode by dialing "Global Inbound Mode feature code + Inbound Mode Suffix" and the current inbound route can be monitored by subscribing a BLF to the Inbound Mode Suffix.
For example, Inbound Default Mode feature code is set to *61 and the Inbound Mode suffix for route 1 is set to 1010. To switch the mode of route 1 to Default Mode, users can dial *611010.
Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating that none of the routes will be affected by the global inbound mode change.
Inbound Route: Inbound Mode BLF Monitoring
Users can assign MPKs and VPKs to monitor and toggle the current global inbound mode of the UCM. To do this, please refer to the following steps:
1. Access the UCM web GUI and navigate to Extension/TrunkInbound Routes.
2. Click on the
button and enable Inbound Multiple Mode.
3. Edit the subscribe number field to the desired BLF value.
Figure 121: Global Inbound Mode
4. Configure the BLF value on a phone's MPK/VPK. As an example, a GXP2140 with the BLF configured will show the Inbound Mode status on its screen once configured. The 777 BLF is lit green, indicating that the current inbound mode is "Default Mode".
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Figure 122: Inbound Mode - Default Mode
5. Pressing the key will toggle the inbound mode to "Mode 1", and the button's color will change to red.
Figure 123: Inbound Mode - Mode 1
Inbound Route: Import/Export Inbound Route
Users can now import and export inbound routes to quickly set up inbound routing on a UCM or to back up an existing configuration. An exported inbound route configuration can be directly imported without needing any manual modifications.
Figure 124: Import/Export Inbound Route
The imported file should be on CSV format and using UTF-8 encoding, the imported file should contain below columns, and each column should be separated by a comma (It is recommended to use Notepad++ for the imported file creation):
Disable This Route: Yes/No. Pattern: Always prefixed with _ CallerID Pattern: Always prefixed with _ Prepend Trunk Name: Yes/No.
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Prepend User Defined Name Enable: Yes/No. Prepend User Defined Name: A string. Alert-info: None, Ring 1, Ring 2... User should enter an Alert-info string following the values we have in
the Inbound route Alert-Info list. Allowed to seamless transfer: [Extension_number] Inbound Multiple Mode: Yes/No. Default Destination: By DID, Extension, Voicemail... User should enter a Default Destination string
following the values we have in the Inbound route Default Destination list. Destination: An Extension number, Ring Group Extension... Default Time Condition. Mode 1: By DID, Extension, Voicemail... User should enter a Default Destination string following the
values we have in the mode 1 Default Destination list. Mode 1 Destination: An Extension number, Ring Group Extension... Mode 1 Time Condition.
FAX with Two Media
The UCM630X supports Fax re-INVITE with multiple codec negotiation. If a Fax re-INVITE contains both T.38 and PCMA/PCMU codec, UCM630X will choose T.38 codec over PCMA/PCMU.
Blacklist Configurations
In the UCM630X, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature and manage the Blacklist by clicking on "Blacklist".
Select the checkbox for "Blacklist Enable" to turn on Blacklist feature for all inbound routes. Blacklist is disabled by default.
Enter a number in "Add Blacklist Number" field and then click "Add" to add to the list. Anonymous can also be added as a Blacklist Number by typing "Anonymous" in Add Blacklist Number field.
To remove a number from the Blacklist, select the number in "Blacklist list" and click on "Clear" button to remove all the numbers on the blacklist.
or click on
User can also export the inbound route blacklist by pressing on
button.
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Figure 125: Blacklist Configuration Parameters
To add blacklist number in batch, click on "Import" to upload blacklist file in csv format. The supported csv format is as below.
Figure 126: Blacklist csv File
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Note: Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature code for "Blacklist Add' (default: *40) and "Blacklist Remove" (default: *41) from an extension. The feature code can be configured under Web GUICall FeaturesFeature Codes.
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FAX SERVER
The UCM6300 series supports T.30/T.38 Fax and Fax Pass-through. It can convert the received Fax to PDF format and send it to the configured Email address. Fax/T.38 settings can be accessed via Web GUICall FeaturesFAX/T.38. The list of received Fax files will be displayed in the same web page for users to view, retrieve and delete.
Configure Fax/T.38
Click on "Create New Fax Extension". In the popped-up window, fill the extension, name and Email address to send the received Fax to.
Click on "Fax Settings" to configure the Fax parameters.
Click on
to edit the Fax extension.
Click on to delete the Fax extension.
Figure 127: Fax Settings
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Table 68: FAX/T.38 Settings
Enable Error Correction Mode
Configure to enable Error Correction Mode (ECM) for the Fax. The default setting is "Yes".
Maximum Transfer Rate
Configure the maximum transfer rate during the Fax rate negotiation. The possible values are 2400, 4800, 7200, 9600, 12000 and 14400. The default setting is 14400.
Configure the minimum transfer rate during the Fax rate negotiation. The Minimum Transfer Rate possible values are 2400, 4800, 7200, 9600, 12000 and 14000. The default
setting is 2400.
Configure the concurrent fax that can be sent by UCM6300. Two modes "Only" and "More" are supported.
Max Concurrent Sending Fax
Only Under this mode, the UCM6300 allows only single user to send fax at a time.
More Under this mode, the UCM6300 supports multiple concurrent fax sending by the users.
By default, this option is set to "only".
Fax Queue Length
Configure the maximum length of Fax Queue from 6 to 10. The default setting is 6.
User Information in Fax If enabled this this will give users the option to send a special header in SIP fax
Header
messages.
Fax Header Information Adds fax header into the fax file.
Configure the Email address to send the received Fax to if user's Email address cannot be found.
Default Email Address Template Variables
Note: The extension's Email address or the Fax's default Email address needs to be configured in order to receive Fax from Email. If neither of them is configured, Fax will not be received from Email. Fill in the "Subject:" and "Message:" content, to be used in the Email when sending the Fax to the users.
The template variables are:
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${CALLERIDNUM} : Caller ID Number ${CALLERIDNAME} : Caller ID Name ${RECEIVEEXTEN} : The extension to receive the Fax ${FAXPAGES} : Number of pages in the Fax ${VM_DATE} : The date and time when the Fax is received
Send PDF Files Only
If enabled, fax emails will no longer attach TIFF files. Only PDF files will be attached.
Enable Fax Resend
Enables the fax resend option which allow the UCM to keep attempting to send faxes up to a specified amount of times. Additionally, if a fax still fails to send, a Resend button will appear in the File Send Progress list in Value-Added FeaturesFax Sending to allow manual resending.
Max Resend Attempts
Configures the maximum attempts number to resend the fax. Default value is set to 5.
Configures the Fax Resend Frequency. Fax Resend Frequency
Default value is set to 50.
Receiving Fax Example Configuration to Receive Fax from PSTN Line
The following instructions describe how to use the UCM6300 to receive fax from PSTN line on the Fax machine connected to the UCM6300 FXS port.
1. Connect Fax machine to the UCM6300 FXS port. 2. Connect PSTN line to the UCM6300 FXO port. 3. Go to Web GUIExtension/Trunk page. 4. Create or edit the analog trunk for Fax as below.
Fax Detection: Make sure "Fax Detection" option is set to "NO".
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Figure 128: Configure Analog Trunk
5. Go to UCM6300 Web GUIExtension/TrunkExtensions page. 6. Create or edit the extension for FXS port.
Analog Station: Select FXS port to be assigned to the extension. By default, it is set to "None". Once selected, analog related settings for this extension will show up in "Analog Settings" section.
Figure 129: Configure Extension for Fax Machine: FXS Extension
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Figure 130: Configure Extension for Fax Machine: Analog Settings
7. Go to Web GUIExtension/TrunkInbound Routes page. 8. Create an inbound route to use the Fax analog trunk. Select the created extension for Fax machine in step
4 as the default destination.
Figure 131: Configure Inbound Rule for Fax
Now the Fax configuration is done. When there is an incoming Fax calling to the PSTN number for the FXO port, it will send the Fax to the Fax machine.
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Example Configuration for Fax-To-Email The following instructions describe a sample configuration on how to use Fax-to-Email feature on the UCM6300. 1. Connect PSTN line to the UCM6300 FXO port. 2. Go to UCM6300 Web GUICall FeaturesFax/T.38 page. Create a new Fax extension.
Figure 132: Create Fax Extension
3. Go to UCM6300 Web GUIExtension/TrunkAnalog Trunks page. Create a new analog trunk. Please make sure "Fax Detection" is set to "No".
4. Go to UCM6300 Web GUIExtension/TrunkInbound Routes page. Create a new inbound route and set the default destination to the Fax extension.
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Figure 133: Inbound Route to Fax Extension
5. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number will be converted to PDF+Tiff file and sent to the extension 7200 and email address fax@domain.local as attachment.
Note: In order for the file to be sent to the email address configured on the external extension, please make sure that the email settings are well configured. Please refer to [Email Settings] section.
Figure 134: List of Fax Files
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FAX Sending
Besides the support of Fax machines, The UCM6300 supports also sending Fax via Web GUI access. This feature can be found on Web GUIValue-added FeaturesFax Sending page. To send fax, pre-setup for analog trunk and outbound route is required. Please refer to [ANALOG TRUNKS], [VOIP TRUNKS] and [Outbound Routes] sections for configuring analog trunk and outbound route.
After making sure analog trunk or VoIP Trunk is setup properly and UCM6300 can reach out to PSTN numbers via the trunk, on Fax Sending page, enter the fax number and upload the file to be faxed. Then click on "Send" to start. The progress of sending fax will be displayed in Web GUI. Users can also view the sending history is in the same web page.
Figure 135: Fax Sending in Web GUI
After that you can see the ongoing sending operation on the progress bar.
Figure 136: Fax Send Progress
Note: Only A3, A4, and B4 paper sizes are supported for the Fax Sending.
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AUDIO CONFERENCE
The UCM630X supports conference room allowing multiple rooms used at the same time. The conference room configurations can be accessed under Web GUICall FeaturesAudio Conference. In this page, users could create, edit, view, invite, manage the participants, and delete conference rooms. The conference room status and conference call recordings (if recording is enabled) will be displayed in this web page as well.
The UCM admin can create multiple audio conference rooms for users to dial in.
UCM630x series Number of audio conference room
UCM6301
3
UCM6302
8
UCM6304
15
UCM6308
25
Participant limit 75 150 200 300
Conference Room Configurations
Click on "Add" to add a new conference room.
Click on
to edit the conference room.
Click on
to delete the conference room.
Extension Privilege Password
Host Password
Table 69: Conference Room Configuration Parameters
Configure the conference number for the users to dial into the conference. Note: Up to 64 characters.
Please select the permission level for outgoing calls. When configured, the users who would like to join the conference call must enter this password before accessing the conference room. Note: The password must be at least 4 characters. Configure the password to join the conference room as Moderator. Conference Moderator can manage the conference call via IVR (if "Enable Caller Menu" is enabled) as well as invite other parties to join the conference by dialing "0" (permission required from the invited party) or "1" (permission not required from the invited party) during the conference call.
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Enable Caller Menu Record Conference
Note: If the Moderator Password field is left blank, moderator functionality will not be available for the meeting.
If enabled, conference participant could press the * key to access the conference room menu. The default setting is "No".
If enabled, the calls in this conference room will be recorded automatically in a .wav format file. All the recording files will be displayed and can be downloaded in the conference web page. The default setting is "No".
Kicking Warning Interval
If there is only one participant in a conference room, a kick warning prompt will play at the configured interval. If no input from the participant is received after the prompt, he will be automatically kicked out of the conference. The valid range is 1-60 minutes.
Wait For Moderator
If enabled, the participants will not hear each other until the host joins the conference. The default setting is "No". Note: If "Quiet Mode" is enabled, the voice prompt for "Wait For Admin" will not be announced.
Allow User Invite
If enabled, users could press 0 to invite other users (with the users' permission) or press 1 to invite other users (without the user's permission) to join the conference. The default setting is "No". Note: Conference administrator can always invite other users without enabling this option.
Announce Callers
Quiet Mode
Play Hold Music Custom Music On Hold
If enabled, the caller will be announced to all conference participants when there the caller joins the conference. The default setting is "No". Note: "Quiet Mode" and "Announce Callers" cannot be enabled at the same time.
If enabled, if there are users joining or leaving the conference, voice prompt or notification tone will not be played. The default setting is "No". Note: "Quiet Mode" and "Announce Callers" cannot be enabled at the same time.
If enabled, the UCM630X will play Hold music when there is only one user in the conference. The default setting is "No".
Select the music on hold class to be played in conference call. Music On Hold class can be set up under Web GUIPBX SettingsMusic On Hold.
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If enabled, the invitation from Web GUI for a conference room with password will Skip Authentication
skip the authentication for the invited users. The default setting is "No".
Conference Settings contains the following options:
Table 70: Conference Settings
Enable Talk detection
If enabled, the AMI will send the corresponding event when a user starts or ends talking.
DSP Talking Threshold
The time in milliseconds of sound above what the dsp has established as base line silence for a user before a user is considered to be talking. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 200.
DSP Silence Threshold
The time in milliseconds of sound falling within the what the dsp has established as base line silence before a user is considered to be silent. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 2500.
Users can check the talking Caller IDs in conference control page (UCM WebUICall FeaturesConference). The image will move up and down when the user is talking.
Figure 137: Conference
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Conference Call Operations Join a Conference Call
Users could dial the conference room extension to join the conference. If password is required, enter the password to join the conference as a normal user, or enter the admin password to join the conference as administrator.
Invite Other Parties to Join Conference
When using the UCM630X conference room., there are two ways to invite other parties to join the conference.
Invite from Web GUI.
For each conference room in UCM630X Web GUICall FeaturesConference, there is an icon
for
option "Invite a participant". Click on it and enter the number of the party you would like to invite. Then click on "Add". A call will be sent to this number to join it into the conference.
Figure 138: Conference Invitation from Web GUI
Invite by dialing 0 or 1 during conference call.
A conference participant can invite other parties to the conference by dialing from the phone during the conference call. Please make sure option "Enable User Invite" is turned on for the conference room first. Enter 0 or 1 during the conference call. Follow the voice prompt to input the number of the party you would like to invite. A call will be sent to this number to join it into the conference.
0: If 0 is entered to invite other party, once the invited party picks up the invitation call, a permission will be asked to "accept" or "reject" the invitation before joining the conference. 1: If 1 is entered to invite other party, no permission will be required from the invited party.
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Note: Conference administrator can always invite other parties from the phone during the call by entering 0 or 1. To join a conference room as administrator, enter the admin password when joining the conference. A conference room can have multiple administrators. ---------------------------------------------------------------------------------------------------------------------------------------------------
During the Conference
During the conference call, users can manage the conference from Web GUI or IVR.
Manage the conference call from Web GUI.
Log in UCM630X Web GUI during the conference call, the participants in each conference room will be listed.
1. Click on
to kick a participant from the conference.
2. Click on to mute the participant.
3. Click on to lock this conference room so that other users cannot join it anymore.
4. Click on
to invite other users into the conference room.
5. Click on
to Invite meeting rooms or invite contact groups.
Manage the conference call from IVR.
If "Enable Caller Menu" is enabled, conference participant can input * to enter the IVR menu for the conference. Please see options listed in the table below.
Table 71: Conference Caller IVR Menu
Conference Administrator IVR Menu
1
Mute/unmute yourself.
2
Lock/unlock the conference room.
3
Kick the last joined user from the conference.
4
Decrease the volume of the conference call.
5
Decrease your volume.
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6
Increase the volume of the conference call.
7
Increase your volume.
More options. 1: List all users currently in the conference call. 2: Kick all non-Administrator participants from the conference call. 8 3: Mute/Unmute all non-Administrator participants from the conference call. 4: Record the conference call. 8: Exit the caller menu and return to the conference.
Conference User IVR Menu
1
Mute/unmute yourself.
4
Decrease the volume of the conference call.
5
Decrease your volume.
6
Increase the volume of the conference call.
7
Increase your volume.
8
Exit the caller menu and return to the conference.
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Note: When there is participant in the conference, the conference room configuration cannot be modified. -------------------------------------------------------------------------------------------------------------------------------------------------
Google Service Settings Support
UCM630X now supports Google OAuth 2.0 authentication. This feature is used for supporting UCM630X conference scheduling system. Once OAuth 2.0 is enabled, UCM630X conference system can access Google calendar to schedule or update conference.
Google Service Settings can be found under Web GUICall FeaturesConferenceGoogle Service SettingsGoogle Service Settings.
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Figure 139: Google Service SettingsOAuth2.0 Authentication
If you already have OAuth2.0 project set up on Google Developers web page, please use your existing login credential for "OAuth2.0 Client ID" and "OAuth2.0 Client Secret" in the above figure for the UCM630X to access Google Service.
If you do not have OAuth2.0 project set up yet, please following the steps below to create new project and obtain credentials:
1. Go to Google Developers page https://console.developers.google.com/start Create a New Project in Google Developers page.
Figure 140: Google ServiceNew Project
2. Enable Calendar API from API Library.
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3. Click "Credentials" on the left drop down menu to create new OAuth2.0 login credentials.
Figure 141: Google ServiceCreate New Credential
4. Use the newly created login credential to fill in "OAuth2.0 Client ID" and "OAuth2.0 Client Secret". 5. Click "Get Authentication Code" to obtain authentication code from Google Service.
Figure 142: Google ServiceOAuth2.0 Login
6. Now UCM630X is connected with Google Service.
You can also configure the Status update, which automatically refresh your Google Calendar with the configured time (m). Note: Zero means disable.
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Conference Schedule
Conference Schedule can be found under UCM630X Web GUI Call Features Audio Conference Conference Schedule. Users can create, edit, view, and delete a Conference Schedule.
Click on "Schedule Meeting" to add a new Conference Schedule. Click on the scheduled conference to edit or delete the event.
After the user configures UCM630X with Google Service Settings [Google Service Settings Support] and enables Google Calendar for Conference Schedule, the conference schedule on the UCM630X can be synchronized with Google Calendar for authorized Google account.
Schedule Options
Table 72: Conference Schedule Parameters
Conference Subject
Configure the name of the scheduled conference. Letters, digits, Other special characters are also supported. such as #%&@*=
Conference Room
Select a conference room for this scheduled conference.
Password
Configure the conference login password.
Time Zone
Select the conference tome zone.
Host Password
Wait For Moderator
Host Schedule Time Meeting Duration Description Repeat Invitees Enable Google Calendar
Remote Conference
Configure Host Password. If enabled, conference participants will not hear each other until the host joins the conference. Note: If Quiet Mode is enabled, the voice prompt for this option will not be played. Configure Host. Set the beginning date and duration of this scheduled conference. Please be aware to avoid time conflicts in the same conference room. The maximum allowed meeting duration that can be set is 8 hour(s). The description of scheduled conference. Choose when to repeat a scheduled conference. Local extensions, remote extensions, and special extensions are supported. Select this option to sync scheduled conference with Google Calendar. Note: Google Service Setting OAuth2.0 must be configured on the UCM630X. Please refer to section [Google Service Settings Support].
Invite a remote conference.
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Conference Room Options
Enable Caller Menu
If this option is enabled, conference participants will be able to access conference room menu by pressing the * key.
Record Conference Quiet Mode Allow User Invite Announce Callers
If this option is enabled, conference call will be recorded in .wav format. The recorded file can be found from Conference page.
If this option is enabled, the notification tone or voice prompt for joining or leaving the conference will not be played. Note: Option "Quiet Mode" and option "Announce Caller" cannot be enabled at the same time.
If this option is enabled, the user can: Press `0' to invite others to join the conference with invited party's permission Press `1' to invite without invited party's permission Press `2' to create a multi-conference room to another conference room Press `3' to drop all current multi-conference rooms.
Note: Conference Administrator is always allowed to access this menu.
If this option is enabled, when a participant joins the conference room, participant's name will be announced to all members in the conference room. Note: Option "Quiet Mode" and option "Announce Caller" cannot be enabled at the same time.
Play Hold Music
If this option is enabled, UCM630X will play Hold Music while there is only one participant in the conference room, or the conference is not yet started.
Custom Music On Hold
Custom Music On Hold.
Skip Authentication
If this option is enabled, the invitation from Web GUI via a trunk with password will not require authentication. Note: Please be aware of the potential security risks when turning on this option.
Once the Conference Schedule is configured, scheduled conference will be displayed as below figure.
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Figure 143: Conference Schedule
Once the conference room is scheduled, at the scheduled conference time, UCM630X will send INVITE to the extensions that have been selected for conference.
Once the conference starts, it will be displayed under Unstarted Meeting with an "Ongoing" status, as displayed below.
Figure 144: Conference Scheduled-Ongoing
Once the conference is finished, the conference will be displayed under Historical meeting as below:
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Figure 145: Conference Schedule-Completed
In addition, once the meeting ends, the system will send a meeting report email to the host including PDF file where he/she can view the meeting, participant information, device type and trend graph of participant levels. Note: Please make sure that outbound route is properly configured for remote extensions to join the conference.
Contact Group
The UCM630X allows users to define a group of contacts and attribute it to a conference room. When a designed extension calls the conference room, the UCM will make a call to all the other numbers on the contact groups to join the conference.
Figure 146: Contact Group
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Conference Recordings
The UCM630X allows users to record the conference call and retrieve the recording from Web GUICall Features Audio ConferenceConference Recordings.
To record the conference call, when the conference room is in idle, enable "Record Conference" from the conference room configuration dialog. Save the setting and apply the change. When the conference call starts, the call will be automatically recorded in .wav format.
The recording files will be listed as below once available. Users could click on to download the recording or click on to delete the recording. Users could also delete all recording files by clicking on "Delate All Recording Files" or delete multiple recording files at once by clicking on "Delete" after selecting the recording files.
Figure 147: Conference Recordings
Conference Call Statistics
Conference reports will now be generated after every conference. These reports can be exported to a .CSV file for offline viewing. The conference report page can be accessed by clicking on the Call Statistics button on the main Conference page.
Figure 148: Conference Call Statistics
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Figure 149: Conference Report on Web Figure 150: Conference Report on CSV
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VIDEO CONFERENCE
With the UCM you can easily create, schedule, manage, and join video conference calls, from your desktop or laptop computer. UCM Video conferencing uses WebRTC technology, so all the participants don't have to download and install any additional software or plugins. UCM Video Conferencing must be enabled by the administrator for the concerned extensions. The video conference configurations can be accessed under Web GUICall FeaturesVideo Conference. In this page, users could enable, set the Basic setting, create, edit, view, manage, delete conference rooms, and edit the Conference Schedule. Below are the UCM video conference specifications supported for each model:
UCM630x series Number of video conference room
UCM6301
4
UCM6302
6
UCM6304
8
UCM6308
10
Participant limit 20 30 60 80
Video Conference
Click on "Add" to add a new conference room. Click on to edit the conference room. Click on to delete the conference room.
Extension Password
Table 73: Video Conference room Configuration Parameters
Configure the conference number for the users to dial into the conference. Note: Up to 64 characters. When configured, the users who would like to join the conference call must enter this password before accessing the conference room.
Note: Only digits are allowed. The password has to be at least 4 characters. All repetitive and
sequential digits (e.g., 0000, 1111, 1234 and 2345) or common digits (e.g., 111222 and 321321) are not allowed.
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Host Password Privilege Allow User Invite
Configure the Host password. Please select the permission level for outgoing calls. If enabled, participants can invite other users to the video conference.
Conference Settings contains the following options:
Table 74: Conference Settings
Enable Talk detection
If enabled, the AMI will send the corresponding event when a user starts or ends talking.
DSP Talking Threshold
The time in milliseconds of sound above what the dsp has established as base line silence for a user before a user is considered to be talking. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 200.
DSP Silence Threshold
The time in milliseconds of sound falling within the dsp has established as base line silence before a user is considered to be silent. This value affects several operations and should not be changed unless the impact on call quality is fully understood, the default value is 2500.
Max Number of Video Feeds Set the maximum number of video feeds supported per conference room.
Audio Codec Preference
Configures the preferred codecs for temporary accounts such as conference participants who joined via link.
Packet Loss Retransmission Packet Loss Retransmission configuration for temporary accounts (conference participants without registered extensions who entered the conference via link)
Jitter Buffer
Select jitter buffer method for temporary accounts such as conference participants who joined via link. Disable: Jitter buffer will not be used. Fixed: Jitter buffer with a fixed size (equal to the value of "Jitter Buffer Size") Adaptive: Jitter buffer with an adaptive size that will not exceed the value of "Max Jitter Buffer"). NetEQ: Dynamic jitter buffer via NetEQ.
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Conference Schedule
Conference Schedule can be found under UCM Web GUI Call FeaturesVideo Conference Conference Schedule. Users can create, edit, view, and delete a Conference Schedule. Click on "Add" to add a new Conference Schedule. Click on the scheduled conference to edit or delete the event.
Schedule Options Conference Subject Conference Room Password
Time Time Zone Host Host password Repeat
Email Reminder (m)
Allow User Invite Call Participants Invitees Description
Table 75: Video Conference Schedule Parameters
Configure the name of the scheduled conference. Letters, digits, _ and are allowed. Select a conference room for this scheduled conference. Configure Conference login password.
Set the beginning date and duration of this scheduled conference. Please be aware to avoid time conflicts in the same conference room. Configure the time zone Configure the conference's host Configure the conference's host password Choose when to repeat a scheduled conference. Email reminders will be sent out x minutes prior to the start of the conference. Valid range is 5-1440. 60 is the default value. 0 indicates not to send out email reminders for the conference. If enabled, participants can invite other users to the video conference. If enabled, invited participants will be called when the meeting starts. Select the participants to invite to the conference. Enter either extension numbers or email addresses. Set a description of scheduled conference.
Once created, at the scheduled conference time, UCM630X will send INVITE to the extensions that have been selected for conference.
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Figure 151: Video Conference Schedule
Once the conference starts, it will be displayed under Unstarted Meeting with an "Ongoing" status, as displayed below.
Figure 152: Video Conference Scheduled-Ongoing
Once the conference is finished, the conference will be displayed under Historical meeting as below:
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Figure 153: Video Conference Scheduled-Completed
In addition, once the meeting ends, the system will send a meeting report email to the host including PDF file where he/she can view the meeting, participant information, device type and trend graph of participant levels
Notes: Video conferencing can be resource-intensive and may cause performance issues with the UCM when used. To ensure the best experience, please use Google Chrome (v67 or higher) or Mozilla Firefox (v60).
Wave WebRTC Video Calling & Conferencing
Web audio and video calls and conferencing can now be achieved through the UCM's new WebRTC page. To get started with this new feature, please make sure to:
1. Navigate to Value-Added Features WebRTC and enable WebRTC support.
Figure 154: Enabling WebRTC Feature
2. Enable the WebRTC on the extensions that would use this feature under Extension / Trunk Extensions by editing the concerned extensions.
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Figure 155: Enabling WebRTC on Extensions
The UCM offers the possibility to login to an extension via Grandstream Wave Portal using user portal password in addition to SIP registration password, where it offers a sleek interface to host conferences, receive email reminders for scheduled conferences, manage contacts, initiate calls, call transfer, chat functionality and more. Access the page by entering the UCM's server address and port which is 8090. (e.g. https://my.ucm.com:8090).
Figure 156: Grandstream Wave Interface
For more details about the Wave Web, please refer to the following guides: http://www.grandstream.com/sites/default/files/Resources/GS_Wave_Web_administration_guide.pdf http://www.grandstream.com/sites/default/files/Resources/GS_Wave_Web_user_guide.pdf
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Grandstream Wave is also available in Androind/IOS version where it can be downloaded from Google Play or AppStorre. For more details, please refer to the following guide: http://www.grandstream.com/sites/default/files/Resources/GSWave_mobile_app_User_Guide.pdf Desktop version is also supported in Windows and MacOS and can be downloaded from: https://fw.gdms.cloud/wave/download/
Figure 157: Download Wave Client
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IVR
Configure IVR
IVR configurations can be accessed under the UCM630X Web GUICall FeaturesIVR. Users could create, edit, view, and delete an IVR. Click on "Add" to add a new IVR. Click on to edit the IVR configuration. Click on to delete the IVR.
Figure 158: Create New IVR
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Basic Settings Name Extension Dial Trunk Auto Record
Permission
Dial Other Extensions
IVR Black/Whitelist Internal Black/Whitelist External Black/Whitelist Replace Display Name Return to IVR Menu Alert Info
Table 76: IVR Configuration Parameters
Configure the name of the IVR. Letters, digits, _ and - are allowed. Enter the extension number for users to access the IVR. If enabled, all callers to the IVR can use trunk. The permission must be configured for the users to use the trunk first. The default setting is "No". If enabled, calls to this IVR will automatically be recorded. Assign permission level for outbound calls if "Dial Trunk" is enabled. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". If the user tries to dial outbound calls after dialing into the IVR, the UCM630X will compared the IVR's permission level with the outbound route's privilege level. If the IVR's permission level is higher than (or equal to) the outbound route's privilege level, the call will be allowed to go through. This controls the destination that can be reached by the external caller via the inbound route. The available destinations are: Extension Conference Call Queue Ring Group Paging/Intercom Groups Voicemail Groups Dial by Name All If enabled only numbers inside of the Whitelist or outside of the Blacklist can be called from IVR.
Contain numbers, either of Blacklist or Whitelist.
This feature can be used only when Dial Trunk is enabled, it contains external numbers allowed or denied calling from the IVR, the allowed format is the following: Number1, number2, number3...
If enabled, the UCM will replace the caller display name with IVR name.
If enabled and if a call to an extension fails, the caller will be redirected to the IVR menu. When present in an INVITE request, the alert-Info header field specifies and alternative ring tone to the UAS.
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Prompt
Select an audio file to play as the welcome prompt for the IVR. Click on "Prompt" to add additional audio file under Web GUIPBX SettingsVoice PromptCustom Prompt.
Digit Timeout
Configure the timeout between digit entries. After the user enters a digit, the user needs to enter the next digit within the timeout. If no digit is detected within the timeout, the UCM630X will consider the entries complete. The default timeout is 3s.
Response Timeout
After playing the prompts in the IVR, the UCM630X will wait for the DTMF entry within the timeout (in seconds). If no DTMF entry is detected within the timeout, a timeout prompt will be played. The default setting is 10 seconds.
Response Timeout Prompt
Select the prompt message to be played when timeout occurs.
Invalid Input Prompt
Select the prompt message to be played when an invalid extension is pressed.
Response Timeout Prompt Repeats
Configure the number of times to repeat the prompt if no DTMF input is detected. When the loop ends, it will go to the timeout destination if configured, or hang up. The default setting is 3.
Invalid Input Prompt Repeats
Configure the number of times to repeat the prompt if the DTMF input is invalid. When the loop ends, it will go to the invalid destination if configured, or hang up. The default setting is 3.
Language
Select the voice prompt language to be used for this IVR. The default setting is "Default" which is the selected voice prompt language under Web GUIPBX SettingsVoice PromptLanguage Settings. The dropdown list shows all the current available voice prompt languages on the UCM630X. To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under Web GUIPBX SettingsVoice PromptLanguage Settings.
Key Pressing Events
Key Press Event: Press 0 Press 1 Press 2 Press 3 Press 4 Press 5 Press 6 Press 7 Press 8 Press 9
Select the event for each key pressing for 0-9, *, Timeout and Invalid. The event options are: Extension Voicemail Conference Rooms Voicemail Group IVR Ring Group Queues Page Group Custom Prompt
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Press *
Timeout Invalid
Hangup DISA Dial by Name External Number Callback When exceeding the number of defined answer timeout, IVR will enter the configured event when timeout. If not configured, then it will Hangup.
Configure the destination when the Invalid Repeat Loop is done.
Figure 159: Key Pressing Events
Black/White List in IVR
In some scenarios, the IPPBX administrator needs to restrict the extensions that can be reached from IVR. For example, the company CEO and directors prefer only receiving calls transferred by the secretary, some special extensions are used on IP surveillance end points which should not be reached from external calls via
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IVR for privacy reason. UCM has now added blacklist and whitelist in IVR settings for users to manage this. Note: up to 500 extensions are allowed on the back/whitelist.
To use this feature, log in UCM Web GUI and navigate to Call FeaturesIVRCreate/Edit IVR: IVR Black/Whitelist.
If the user selects "Blacklist Enable" and adds extension in the list, the extensions in the list will not be allowed to be reached via IVR.
If the user selects "Whitelist Enable" and adds extension in the list, only the extensions in the list can be allowed to be reached via IVR.
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Figure 160: Black/Whitelist
Create Custom Prompt
To record new IVR prompt or upload IVR prompt to be used in IVR, click on "Upload Audio File" next to the "Welcome Prompt" option and the users will be redirected to Custom Prompt page. Or users could go to Web GUIPBX SettingsVoice PromptCustom Prompt page directly.
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Figure 161: Click on Prompt to Create IVR Prompt
Once the IVR prompt file is successfully added to the UCM630X, it will be added into the prompt list options for users to select in different IVR scenarios.
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LANGUAGE SETTINGS FOR VOICE PROMPT
The UCM630X supports multiple languages in Web GUI as well as system voice prompt. Currently, there are 16 languages supported in system voice prompt: English (United States), Arabic, Chinese, Dutch, English (United Kingdom), French, German, Greek, Hebrew, Italian, Polish, Portuguese, Russian, Spanish, Catalan, Swedish and Turkish.
English (United States) and Chinese voice prompts are built in with the UCM630X already. The other languages provided by Grandstream can be downloaded and installed from the UCM630X Web GUI directly. Additionally, users could customize their own voice prompts, package them and upload to the UCM630X.
Language settings for voice prompt can be accessed under Web GUIPBX SettingsVoice PromptLanguage Settings.
Download and Install Voice Prompt Package
To download and install voice prompt package in different languages from UCM630X Web GUI, click on "Add Voice Prompt Package" button.
Figure 162: Language Settings for Voice Prompt
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A new dialog window of voice prompt package list will be displayed. Users can see the version number (latest version available V.S. current installed version), package size and options to upgrade or download the language.
Figure 163: Voice Prompt Package List
Click on to download the language to the UCM630X. The installation will be automatically started once the downloading is finished.
Figure 164: New Voice Prompt Language Added
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A new language option will be displayed after successfully installed. Users then could select it to apply in the UCM630X system voice prompt or delete it from the UCM630X.
Customize Specific Prompt
On the UCM630X, if the user needs to replace some specific customized prompt, the user can upload a single specific customized prompt from Web GUIPBX SettingsVoice PromptLanguage Settings and click on "Upload" instead of the entire language pack.
Figure 165: Upload Single Voice Prompt for Entire Language Pack
Username Prompt Customization
There are two ways to customize/set new username prompt:
Upload Username Prompt File from Web GUI
1. First, Users should have a pre-recorded file respecting the following format:
PCM encoded / 16 bits / 8000Hz mono. In .tar/.tar.gz/.tgz format File size under 30M. Filename must be set as the extension number with 18 characters max. For example, the recorded
file name 1000.wav will be used for extension 1000.
2. Go under web GUI PBX Settings Voice Prompt Username Prompt and click on "Upload" button.
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3. Select the recorded file to upload it and press Save and Apply Settings.
Click on
to record again the username prompt.
Click on to play recorded username prompt.
Select username prompts and press the button "Delete" .
Record Username via Voicemail Menu
to delete specific file or select multiple files for deletion using
The second option to record username is using voicemail menu, please follow below steps:
Dial *98 to access the voicemail After entering the desired extension and voicemail password, dial "0" to enter the recordings menu and
then "3" to record a name.
Another option is that each user can record their own name by following below steps: The user dials *97 to access his/her voicemail After entering the voicemail password, the user can press "0" to enter the recordings menu and then "3" to record his name.
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VOICEMAIL
Configure Voicemail
If the voicemail is enabled for UCM630X extensions, the configurations of the voicemail can be globally set up and managed under Web GUICall FeaturesVoicemail.
Figure 166: Voicemail Settings
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Max Greeting Time (s) Dial `0' For Operator Operator Type Operator Extension Max Messages Per Folder
Max Message Time
Min Effective Message Time
Announce Message Caller-ID Announce Message Duration Play Envelope Play Most Recent First
Table 77: Voicemail Settings
Configure the maximum number of seconds for the voicemail greeting. The default setting is 60 seconds. If enabled, the caller can press 0 to exit the voicemail application and connect to the configured operator's extension. Configure the operator type; either an extension or a ring group. Select the operator extension, which will be dialed when users press 0 to exit voicemail application. The operator extension can also be used in IVR. Configure the maximum number of messages per folder in users' voicemail. The valid range 10 to 1000. The default setting is 50. Select the maximum duration of the voicemail message. The message will not be recorded if the duration exceeds the max message time. The default setting is 15 minutes. The available options are: 1 minute 2 minutes 5 minutes 15 minutes 30 minutes Unlimited Configure the minimum duration (in seconds) of a voicemail message. Messages will be automatically deleted if the duration is shorter than the Min Message Time. The default setting is 3 seconds. The available options are:
No minimum 1 second 2 seconds 3 seconds 4 seconds 5 seconds
Note: Silence and noise duration are not counted in message time. If enabled, the caller ID of the user who has left the message will be announced at the beginning of the voicemail message. The default setting is "No". If enabled, the message duration will be announced at the beginning of the voicemail message. The default setting is "No". If enabled, a brief introduction (received time, received from, and etc.) of each message will be played when accessed from the voicemail application. The default setting is "Yes". If enabled, it will play the most recent message first.
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Allow User Review
Voicemail Remote Access
Forward Voicemail to Peered UCMs Voicemail Password Format
If enabled, users can review the message following the IVR before sending. If enabled, external callers routed by DID and reaching VM will be prompted by the UCM with 2 options:
Press 1 to leave a message. To leave a message for the extension reached by DID.
Press 2 to access voicemail management system. This will allow caller to access any extension VM after entering extension number and its VM password.
Note: This option applies to inbound call routed by DID only. The default setting is "Disabled". Enables the forwarding of voicemail to remote extensions on peered SIP trunks. The default setting is "Disabled".
Configures the default voicemail password that will be used when an extension is reset.
Warning: WAV files take up significantly more storage space than GSM files.
Note: Resetting an extension will reset Voicemail Password, Send Voicemail to Email, and Keep Voicemail after Emailing values to default. Previous custom voicemail prompts and messages will be deleted.
Access Voicemail
If the voicemail is enabled for UCM630X extensions, the users can dial the voicemail access number (by default *97) to access their extension's voicemail. The users will be prompted to enter the voicemail password and then can enter digits from the phone keypad to navigate in the IVR menu for different options.
Otherwise the user can dial the voicemail access code (by default *98) followed by the extension number and password in order to access to that specific extension's voicemail.
Main Menu 1 New messages
Table 78: Voicemail IVR Menu
Sub Menu 1
Sub Menu 2
3 - Advanced options
1 - Send a reply
2 - Call the person who sent this message
3 - Hear the message envelop
4 - Leave a message
* - Return to the main menu
5 - Repeat the current message
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2 Change folders 3 Advanced options 0 Mailbox options
7 - Delete this message 8 - Forward the message to another user 9 Save * - Help # - Exit 0 - New messages 1 - Old messages 2 - Work messages 3 - Family messages 4 - Friend messages # - Cancel 1 - Send a reply 2 - Call the person who sent this message 3 - Hear the message envelop 4 - Leave a message * - Return to the main menu 1 - Record your unavailable message
2 - Record your busy message
3 - Record your name
4 - Record temporary greeting
5 - Change your password * - Return to the main menu
1 - Accept this recording 2 - Listen to it 3 - Re-record your message 1 - Accept this recording 2 - Listen to it 3 - Re-record your message 1 - Accept this recording 2 - Listen to it 3 - Re-record your message 1 - Accept this recording 2 - Listen to it 3 - Re-record your message
Leaving Voicemail
If an extension has voicemail enabled under basic settings "Extension/Trunk Extensions Basic Settings"
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and after a ring timeout or user not available, the caller will be automatically redirected to the voicemail in order to leave a message on which case they can press # in order to submit the message.
In case if the caller is calling from an internal extension, they will be directly forwarded to the extension's voicemail box. But if the caller is calling from outside the system and the incoming call is routed by DID to the destination extension, then the caller will be prompted with the choice to either press1 to access voicemail management or press 2 to leave a message for the called extension. This feature could be useful for remote voicemail administration.
Voicemail Email Settings
The UCM630X can be configured to send the voicemail as attachment to Email. Click on "Voicemail Email Settings" button to configure the Email attributes and content.
Table 79: Voicemail Email Settings
If enabled, voicemail will be sent to the user's email address. Note: SMTP Send Voicemail to Email
server must be configured to use this option.
Keep Voicemail after Emailing
Enable this option if you want to keep recording files after the Email is sent. The default setting is Enable.
Email Template
Fill in the "Subject:" and "Message:" content, to be used in the Email when sending to the user. The template variables are: \t: TAB ${VM_NAME}: Recipient's first name and last name ${VM_DUR}: The duration of the voicemail message ${VM_MAILBOX}: The recipient's extension ${VM_CALLERID}: The caller ID of the person who has left the message ${VM_MSGNUM}: The number of messages in the mailbox ${VM_DATE}: The date and time when the message is left
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Figure 167: Voicemail Email Settings
Click on "Email Template" button to view the default template as an example.
Configure Voicemail Group
The UCM630X supports voicemail group and all the extensions added in the group will receive the voicemail to the group extension. The voicemail group can be configured under Web GUI Call Features Voicemail Voicemail Group. Click on "Add" to configure the group.
Figure 168: Voicemail Group
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Extension
Name Voicemail Password Email Address Member
Table 80: Voicemail Group Settings
Enter the Voicemail Group Extension. The voicemail messages left to this extension will be forwarded to all the voicemail group members. Configure the Name to identify the voicemail group. Letters, digits, _ and - are allowed.
Configure the voicemail password for the users to check voicemail messages.
Configure the Email address for the voicemail group extension. Select available mailboxes from the left list and add them to the right list. The extensions need to have voicemail enabled to be listed in available mailboxes list.
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RING GROUP
The UCM630X supports ring group feature with different ring strategies applied to the ring group members. This section describes the ring group configuration on the UCM630X.
Configure Ring Group
Ring group settings can be accessed via Web GUICall FeaturesRing Group.
Figure 169: Ring Group
Click on
to add ring group.
Click on
to edit the ring group. The following table shows the ring group configuration parameters.
Click on
to delete the ring group.
Ring Group Name Extension Members LDAP Phonebook
Ring Strategy
Table 81: Ring Group Parameters
Configure ring group name to identify the ring group. Letters, digits, _ and are allowed.
Configure the ring group extension.
Select available users from the left side to the ring group member list on the
right side. Click on
to arrange the order.
Select available remote users from the left side to the ring group member list
on the right side. Click on be enabled first.
to arrange the order. Note: LDAP Sync must
Select the ring strategy. The default setting is "Ring in order".
Ring simultaneously. Ring all the members at the same time when there is incoming call to the ring group extension. If any of the member answers the call, it will stop
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Music On Hold Custom Prompt
Ring Timeout on Each Member Auto Record
Endpoint Call Forwarding Support
ringing.
Ring in order. Ring the members with the order configured in ring group list. If the first member does not answer the call, it will stop ringing the first member and start ringing the second member.
Select the "Music On Hold" Class of this Ring Group, "Music On Hold" can be managed from the "Music On Hold" panel on the left.
This option is to set a custom prompt for a ring group to announce to caller. Click on `Prompt', it will direct the users to upload the customized voice prompts. Note: Users can also refer to the page PBX SettingsVoice PromptCustom Prompt, where they could record new prompt or upload prompt files.
Configure the number of seconds to ring each member. If set to 0, it will keep ringing. The default setting is 60 seconds.
Note: The actual ring timeout might be overridden by users if the phone has ring timeout settings as well.
If enabled, calls on this ring group will be automatically recorded. The default setting is No. The recording files can be accessed from Web GUICDRRecording Files. This allows the UCM to work with endpoint-configured call forwarding settings to redirect calls to ring group. For example, if a member wants to receive calls to the ring group on his mobile phone, he will have to set his endpoint's call forwarding settings to his mobile number. By default, it is disabled.
However, this feature has the following limitations:
This feature will work only when call forwarding is configured on endpoints, not on the UCM.
If the forwarded call goes through an analog trunk, and polarity reversal is disabled, the other ring group members will no longer receive the call after it is forwarded.
If the forwarded call goes through a VoIP trunk, and the outbound route for it is PIN-protected and requires authentication, the other ring group members will no longer receive the call after it is forwarded.
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Replace Display Name Skip Busy Agent Enable Destination
Default Destination
If the forwarded call hits voicemail, the other ring group members will no longer receive the call.
If enabled, the UCM will replace the caller display name with the Ring Group name the caller know whether the call is incoming from a direct extension or a Ring Group.
If enabled, skip busy agents regardless of call waiting settings.
If enabled, users could select extension, voicemail, ring group, IVR, call queue, voicemail group as the destination if the call to the ring group has no answer. Secret and Email address are required if voicemail is selected as the destination. The call would be routed to this destination if no one in this ring group answers the call. Note: Users can now set the voicemail of ring groups as routing destinations and IVR key press event destinations and to do so ring group must have their Default Destination set to Voicemail with Ring Group Extensions.
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Figure 170: Ring Group Configuration
Remote Extension in Ring Group
Remote extensions from the peer trunk of a remote UCM630X can be included in the ring group with local extension. An example of Ring Group with peer extensions is presented in the following:
1. Creating SIP Peer Trunk between both UCM630X _A and UCM630X _B. SIP Trunk can be found under Web GUIExtension/TrunkVoIP Trunks. Also, please configure their Inbound/Outbound routes accordingly.
2. Click edit button in the menu , and check if Sync LDAP Enable is selected, this option will allow UCM630X_A update remote LDAP server automatically from peer UCM630X _B. In addition, Sync LDAP Password must match for UCM630X _A and UCM630X _B to sync LDAP contact automatically. Port number
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can be anything between 0~65535, and use the outbound rule created in step 1 for the LDAP Outbound Rule option.
Figure 171: Sync LDAP Server option
3. In case if LDAP server does not sync automatically, user can manually sync LDAP server. Under VoIP Trunks page, click sync button shown in the following figure to manually sync LDAP contacts from peer UCM630X.
Figure 172: Manually Sync LDAP Server
4. Under Ring Groups setting page, click "Add". Ring Groups can be found under Web GUICall FeaturesRing Groups.
5. If LDAP server is synced correctly, Available LDAP Numbers box will display available remote extensions that can be included in the current ring group. Please also make sure the extensions in the peer UCM630X can be included into that UCM630X's LDAP contact.
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Figure 173: Ring Group Remote Extension
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PAGING AND INTERCOM GROUP
Paging and Intercom Group can be used to make an announcement over the speaker on a group of phones. Targeted phones will answer immediately using speaker. The UCM630X paging and intercom can be used via feature code to a single extension or a paging/intercom group. This section describes the configuration of paging/intercom group under Web GUICall FeaturesPaging/Intercom.
Configure Paging/Intercom Group
Click on "Add" to add paging/intercom group.
Click on
to edit the paging/intercom group.
Click on
to delete the paging/intercom group.
Click on "Paging/Intercom Group Settings" to edit Alert-Info Header. This header will be included in the SIP INVITE message sent to the callee in paging/intercom call.
Configure Multicast Paging
Figure 174: Multicast Paging
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Table 82: Multicast Paging Configuration Parameters
Name
Configure paging/intercom group name.
Type
Select "Multicast Paging".
Extension
Configure the paging/intercom group extension.
Multicast IP Address
The allowed multicast IP address range is 224.0.1.0 - 238.255.255.255. Note: This field appears only when "Type" is set to "Multicast Paging".
Specify the maximum call duration in seconds. The default value 0 means no Maximum Call Duration
limit.
Custom Prompt
This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on `Prompt', it will direct the users to upload the customized voice prompts. Note: Users can also refer to the page PBX SettingsVoice PromptCustom Prompt, where they could record new prompt or upload prompt files.
Multicast IP Address
Select which extensions are allowed to use the paging/intercom feature for this paging group.
Port
Specify port for multicast paging. Note: This field appears only when "Type" is set to "Multicast Paging".
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Configure 2-way Intercom
Figure 175: 2-way Intercom
Name
Table 83: 2-way Intercom Configuration Parameters
Configure paging/intercom group name.
Type
Select "2-way Intercom".
Extension
Configure the paging/intercom group extension.
Auto Record
Enable this option to record in WAV format.
If enabled, the UCM will replace the caller display name with Paging/Intercom Replace Display Name
name.
Specify the maximum call duration in seconds. The default value 0 means no Maximum Call Duration
limit.
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Custom Prompt
Members Paging/Intercom Whitelist
This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on `Prompt', it will direct the users to upload the customized voice prompts. Note: Users can also refer to the page PBX SettingsVoice PromptCustom Prompt, where they could record new prompt or upload prompt files. Select available users from the left side to the paging/intercom group member list on the right. Select which extensions are allowed to use the paging/intercom feature for this paging group.
Configure 1-way Paging
Figure 176: 1-way Paging
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Table 84: 1-way Paging Configuration Parameters
Name
Configure paging/intercom group name.
Type
Select "1-way Paging".
Extension
Configure the paging/intercom group extension.
Auto Record
Enable this option to record in WAV format.
If enabled, the UCM will replace the caller display name with Paging/Intercom Replace Display Name
name.
Specify the maximum call duration in seconds. The default value 0 means no Maximum Call Duration
limit.
Custom Prompt
This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on `Prompt', it will direct the users to upload the customized voice prompts. Note: Users can also refer to the page PBX SettingsVoice PromptCustom Prompt, where they could record new prompt or upload prompt files.
Members
Select available users from the left side to the paging/intercom group member list on the right.
Paging/Intercom Whitelist
Select which extensions are allowed to use the paging/intercom feature for this paging group.
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Configure Announcement Paging
Enable Name Type
Custom Prompt
Repeat Date
Figure 177: Announcement Paging
Table 85: Announcement Paging Configuration Parameters
Enable/Disable Announcement Paging. Configure paging/intercom group name.
Select "Announcement Paging" This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on `Prompt', it will direct the users to upload the customized voice prompts. Note: Users can also refer to the page PBX SettingsVoice PromptCustom Prompt, where they could record new prompt or upload prompt files. If enabled, the announcement page will be repeated for the selected weekdays. Configure Announcement Paging Date.
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Time Transmission Method Members
Configure Announcement Paging Time. Configure Announcement Paging transmission method. Unicast: Depending on members selection Multicast: Depending on Multicast IP address and Port Select available users from the left side to the paging/intercom group member list on the right.
Paging/Intercom Group Settings
Figure 178: Page/Intercom Group Settings
The UCM630X has pre-configured paging/intercom feature code. By default, the Paging Prefix is *81 and the Intercom Prefix is *80. To edit page/intercom feature code, click on "Feature Codes" in the "Paging/Intercom Group Settings" dialog. Or users could go to Web GUICall FeaturesFeature Codes directly.
Configure a Scheduled Paging/Intercom
Users can schedule paging/intercom calls by using the Schedule Paging/Intercom page. To schedule, click the Add button on the new page and configure the caller, the group to use, and the time to call out.
Figure 179: Schedule Paging/Intercom page
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Table 86: Schedule Paging / Intercom Settings
Caller
Configure the caller ID for the paging / intercom group.
Paging/Intercom Group Select the paging / intercom group from the list of the available groups.
Start Time
Configure the start time of the scheduled paging / intercom call.
Type Action Status
Select the type for the scheduled paging / intercom call. The available types are: Single time or Daily basis. Default is "Single". Display the action status of the scheduled paging / intercom call.
Figure 180: Creating a scheduled paging/intercom call
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CALL QUEUE
The UCM630X supports call queue by using static agents or dynamic agents. Call Queue system can accept more calls than the available agents. Incoming calls will be held until next representative is available in the system. This section describes the configuration of call queue under Web GUICall FeaturesCall Queue.
Configure Call Queue
Call queue settings can be accessed via Web GUICall FeaturesCall Queue.
Figure 181: Call Queue
UCM630X supports custom prompt feature in call queue. This custom prompt will active after the caller waits for a period of time in the Queue. Then caller could choose to leave a message/ transfer to default extension or keep waiting in the queue.
To configure this feature, please go to UCM Web GUICall FeaturesCall QueueCreate New Queue/Edit QueueQueue Optionsset Enable Destination to Enter Destination with Voice Prompt. Users could configure the wait time with Voice Prompt Cycle.
Click on "Add" to add call queue. Click on to edit the call queue. The call queue configuration parameters are listed in the table below.
Click on to delete the call queue.
Basic Settings Extension Name
Strategy
Table 87: Call Queue Configuration Parameters
Configure the call queue extension number. Configure the call queue name to identify the call queue. Select the strategy for the call queue. Ring All
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Music On Hold
Max Queue Length
Wrapup Time Retry Time Ring Time Auto Record Max Wait Time Welcome Prompt Destination
Ring all available Agents simultaneously until one answers.
Linear Ring agents in the specified order.
Least Recent Ring the agent who has been called the least recently.
Fewest Calls Ring the agent with the fewest completed calls.
Random Ring a random agent.
Round Robin Ring the agents in Round Robin scheduling with memory.
The default setting is "Ring All". Select the Music On Hold class for the call queue.
Note: Music On Hold classes can be managed from Web GUIPBX SettingsMusic On Hold. Configure the maximum number of calls to be queued at once. This number does not include calls that have been connected with agents. It only includes calls not connected yet. The default setting is 0, which means unlimited. When the maximum value is reached, the caller will be treated with busy tone followed by the next calling rule after attempting to enter the queue. Configure the number of seconds before a new call can ring the queue after the last call on the agent is completed. If set to 0, there will be no delay between calls to the queue. The default setting is 10 seconds. Configure the number of seconds to wait before ringing the next agent. Configure the number of seconds an agent will ring before the call goes to the next agent. The default setting is 30 seconds. If enabled, the calls on the call queue will be automatically recorded. The recording files can be accessed in Queue Recordings under Web GUICall FeaturesCall Queue. Configure the timeout after which users will be disconnected from the call queue. The default setting is "60". 0 means unlimited. Note: It is recommended to configure "Wait Time" longer than the "Wrapup Time". If enabled, users can upload an audio file that will be played as an Initial tone when dialing the queue number. Once Max Wait Time has been configured, select to which destination send the calls that have timed out. The default is to "Hang up" the call.
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Destination Prompt Cycle Custom Prompt Destination Advanced Settings - Virtual Queue - Caller
Announcement - Queue Chairman Enable Position Announcement Enable Wait Time Announcement Announcement Interval Enable Agent Login
Leave When Empty
Dial in Empty Queue
Configure the voice prompt cycle (in seconds) of the call queue. Once all agents are busy and the voice prompt will be played, and you can press the appropriate key to transfer to failover destination. When playing a custom prompt, press 1 to transfer to failover destination. Select failover destination to send callers after pressing 1 upon hearing the custom prompt.
Refer to Call Center Settings and Enhancements section for detailed information about these features.
If enabled, the system will inform callers waiting in the queue of their positions in line. If enabled, the estimated wait time for the call to get answered will periodically be announced to the caller. Note: Wait time will not be announced if less than one minute.
The interval at which caller positions and estimated wait times will be announced.
Enables agent login/logout feature for static agents (supported only on GXP21XX phones with firmware higher than 1.0.9.18). Configure whether the callers will be disconnected from the queue or not if the queue has no agent anymore. The default setting is "Strict". Yes
Callers will be disconnected from the queue if all agents are paused or invalid. No
Never disconnect the callers from the queue when the queue is empty. Strict
Callers will be disconnected from the queue if all agents are paused, invalid or unavailable. Configure whether the callers can dial into a call queue if the queue has no agent. The default setting is "No". Yes Callers can always dial into a call queue. No Callers cannot dial into a queue if all agents are paused or invalid.
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Strict Callers cannot dial into a queue if the agents are paused, invalid or unavailable.
Failover Destination
Choose the destination where the call will be directed when the queue is empty or when all the agents are not logged in, here are the destinations that can be configured:
Play Sound. Extension. Voicemail. Queues. Ring Group. Voicemail Group. IVR. External Number.
Enable Agent Login Enabling agent login will cause the dynamic agents to be unavailable.
Queue Chairman
The queue chairman can log into his web portal to operate the queue.
Report Hold Time
If enabled, the UCM630X will report (to the agent) the duration of time of the call before the caller is connected to the agent. The default setting is "No".
If enabled, the UCM will replace the caller display name with the Call Queue name Replace Display Name
so that the caller knows the call is incoming from a Call Queue.
Enable feature codes option for call queue. For example, *83 is used for "Agent Enable Feature Codes
Pause"
Autofill
Configure to enable autofill.
Dynamic Login Password
If enabled, the configured PIN number is required for dynamic agent to log in. The default setting is disabled.
Alert-Info
When present in an INVITE request, the Alert-info header field specifies an alternative ring tone to the UAS.
Agents
Static Agents
Go to "Agents" Tab and Select the available users to be the static agents in the
call queue. Choose from the available users on the left to the static agents list on
the right. Click on
to choose. And use UP and Down arrow to select the
order of the agent within the call queue.
Static Agents limitation:
To guarantee a high level of audio quality with the call queue feature, UCMs will limit the number of static agents allowed to be assigned depending on the UCM model used. If the user attempts to configure the number of static agents to be more than the maximum allowed number, a warning message will appear.
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The following table lists the maximum number of static agents for each UCM model:
UCM Model
UCM6301 UCM6302 UCM6304 UCM6308
Table 88: Static Agent Limitation
Max Static Agents in Call Queue 10 60 100 120
Click on "Global Queue Settings" to configure Agent Login Extension Postfix and Agent Logout Extension Postfix. Once configured, users could log in the call queue as dynamic agent.
Figure 182: Agent Login Settings
For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout Extension Postfix is **, users could dial 6500* to login to the call queue as dynamic agent and dial 6500** to logout from the call queue. Dynamic agent does not need to be listed as static agent and can log in/log out at any time.
Call queue feature code "Agent Pause" and "Agent Unpause" can be configured under Web GUICall FeaturesFeature Codes. The default feature code is *83 for "Agent Pause" and *84 for "Agent Unpause". Note: When dialing the "Agent Pause" feature code, users can specify the reason for it. The following reasons are available: (1) Lunch, (2) Hourly Break, (3) Backoffice, (4) Email, and (5) Wrap.
Queue recordings are shown on the Call Queue page under "Queue Recordings" Tab. Click on to download the recording file in .wav format; click on to delete the recording file. To delete multiple
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recording files by one click, select several recording files to be deleted and click on "Delete Selected Recording Files" or click on "Delete All Recording Files" to delete all recording files.
Call Center Settings and Enhancements
UCM supports light weight call center features including virtual queue and position announcement, allowing the callers to know their position on the call queue and giving them the option to either stay on the line waiting for their turn or activate a callback which will be initiated by the UCM one an agent is free.
To configure call center features, press Following parameters are available:
on an existing call queue and go under the advanced settings tab.
Enable Virtual Queue Virtual Queue Period
Virtual Queue Mode
Virtual Queue Outbound Prefix Enable Virtual Queue Timeout Write Timeout Enable Virtual Queue Position Announcement
Table 89: Call Center Parameters
Enable virtual queue to activate call center features. Configure the time in (s) after which the virtual queue will take effect and the menu will be presented to the caller to choose an option. Default is 20s. Offered to caller after timeout: After the virtual queue period passes, the caller will enter the virtual call queue and be presented with a menu to choose an option, the choices are summarized below: Press * to set current number as callback number. Press 0 to set a callback number different than current caller number. Press # to keep waiting on the call queue.
Triggered on user request: In this mode, the callers can activate the virtual queue by pressing 2, then they will be presented with the menu to choose an option as below: Press * to set current number as callback number. Press 0 to set a callback number different than current caller number. Press # to keep waiting on the call queue.
System will add this prefix to dialed numbers when calling back users.
When this option is enabled and after a caller registers a call back request on the virtual queue. While all the agents are busy, the UCM will call an agent once he/she is idle again, this timeout is used for how long the UCM continues calling the agent and if the agent doesn't answer the call then the callback request will timeout and expire. Configure the virtual queue callback timeout period in seconds. Enable the announcement of the caller's position periodically. Note: Queue position will now be announced to the caller upon entering the queue.
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Position Announcement Interval Enable Virtual Queue Wait Time Announcement Queue Chairman Virtual Queue Welcome Prompt
Configure the period of time in (s) during which the UCM will announce the caller's position in the call queue.
When enabled the UCM will announce the estimated queue wait time to callers if the estimated wait time is longer than 1 minute.
Select the extension to act as chairman of the queue (monitoring).
Click on "Upload Audio File" to upload the VQ welcome prompt.
When enabled, statics agents can conveniently log in and out of a queue by configuring a programmable key on their phones as a shortcut. Notes:
This feature is currently available only for GXP21xx phones on firmware 1.0.9.18 or greater.
Enable Agent Login
After enabling the feature, users need to set the option on GXP21XX phone under "AccountSIP SettingsAdvanced FeaturesSpecial Feature" to "UCM Call Center". A softkey labeled "UCM-CC" will appear on the bottom of the phone's screen.
When this option is enabled, dynamic agent login will be no longer supported.
In case of concurrent registrations, changing agent status on one phone (login/logout) will be reflected on all phones.
Queue Auto fill enhancement:
The waiting callers are connecting with available members in a parallel fashion until there are no more available members or no more waiting callers.
For example, in a call queue with linear method, if there are two available agents, when two callers call in the queue at the same time, UCM will assign the two callers to each of the two available agents at the same time, rather than assigning the second caller to second available agent after the first agent answers the call from the first caller.
Queue Statistics
Along with call center features, users can also gather detailed call queue statistics allowing them to make better changes/decision to manage better the call distribution and handling based on time, agent, and queue. To access call queue statistics, go to Web GUICall FeaturesCall Queue and click on "Call Queue Statistics", the following page will be displayed:
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Figure 183: Call Queue Statistics
By selecting a time interval, administrators can get detailed statistics for agent(s) such as total calls, answered calls etc, as well as for the queue(s) such as ABANDONED CALLS also a detailed information for the queue's call log by clicking on OptionsInformation button and the below window will pop up:
Figure 184: Queue's call log details
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User can download statistics on CSV format by clicking on the "Download", also the statistics can be cleared using "Reset Statistics" button.
The statistics can be automatically sent to a specific email address on a preconfigured Period, this can be done by clicking on "Automatic Download", and user will be directed to below page where he can configure the download period (Day/Week/Month) and the Email where the statistics will be sent (Email settings should be configured correctly):
Figure 185: Automatic Download Settings - Queue Statistics
Significantly more information is now available UCM's queue statistics page. In addition to the information presented in previous firmware, users can now view a call log that displays calls to all agents and queues, a dynamic agent login/logout record, and a pause log. Statistics reports for these new pages can be obtained by pressing the Download button in the top left corner of the Call Queue Statistics page. The reports are in .CSV format and will be packaged into a single tar.gz file upon download.
Agent Details is a call log that shows every call to each individual agent from all queues. The following information is available:
Time the date and time the call was received. Agent the agent that was rung for the call. Queue the queue that the call went to. Caller ID Number the CID of the caller Abandoned indicates whether the call was picked up or not by that specific agent. If the call rang
several agents simultaneously, and this specific agent did not pick up the call, the call will be considered abandoned even if a different agent in the same queue picked it up. Wait Time the amount of time that the call was waiting in queue after dialing in. Talk Time the duration of the call after it was picked up by agent.
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Figure 186: Agent details
Login Record is a report that shows the timestamps of dynamic agent logins and logouts and calculates the amount of time the dynamic agents were logged in. Dynamic agents are extensions that log in and out either via agent login/logout codes (configured in Global Queue Settings page) or by using the GXP21xx call queue softkey. A new record will be created only when an agent logs out. The following information is available:
Agent the extension that logged in and out. Queue the queue that the extension logged in and out of. Login Time the time that the extension logged into the queue. Logout Time the time that the extension logged out of the queue. Login Duration the total length of time that the extension was logged in.
Figure 187: Login Record
Pause Log is a report that shows the times of agent pauses and unpauses and calculates the amount of time that agents are paused. If an agent is part of several queues, an entry will be created for each queue. An entry will only be created after an agent unpauses. The following information is available:
Agent the extension that paused and unpaused. Queue the queue that the agent is in. Pause Time the time that the agent paused. Resume Time the time that the agent unpaused. Pause Duration the total length of time the agent was paused for.
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Figure 188: Pause Log
Switchboard
Switchboard is a Web GUI tool for call queue monitoring and management, admin can access to it from the menu Call FeaturesCall Queue then press "Switchboard". Following page will be displayed:
Figure 189: Switchboard Summary
Page above summarizes the available queues statistics and if one of the queues is clicked the user will be directed to page below:
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Figure 190: Call Queue Switchboard
The table below gives a brief description for the main menus:
Waiting Proceeding
Agents
Table 90: Switchboard Parameters
This menu shows the current waiting calls along with the caller id and the option to hang-up call by pressing on the button. Shows the current established calls along with the caller id and the callee (agent) as well as the option to hang-up, transfer, add conference or barge-in the call. Displays the list of agents in the queue and the extension status (idle, ringing, in use or unavailable) along with some basic call statistics and agent's mode (static or dynamic).
Note: the dashboard will show the number of calls (answered and abandoned) of each agent. For dynamic agents, it will count the number of calls starting from the last login time.
There are three different privilege levels for Call Queue management from the switchboard: Super Admin, Queue Chairman, and Queue Agent.
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Super Admin - Default admin of the UCM. Call queue privileges include being able to view and edit all queue agents, monitor, and execute actions for incoming and ongoing calls for each extension in Switchboard, and generate Call Queue reports to track performance.
Queue Chairman - User appointed by Super Admin to monitor and manage an assigned queue extension via Switchboard. The Queue Chairman can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on "Value-added Features" in the side menu and click on "Call Queue". In the image below, User 1001 is the Queue Chairman appointed to manage Queue Extension 6500 and can see all the agents of the queue in the Switchboard.
Figure 191: Queue Chairman
Queue Agent - User appointed by Super Admin to be a member of a queue extension. A queue agent can log into the UCM user portal with his extension number and assigned user password. To access the Switchboard, click on "Value-added Features" in the side menu and click on "Call Queue". However, a queue agent can view and manage only his own calls and statistics, but not other agents' in the queue extension. In the image below, User 1000 is a queue agent and can see only his own information in the Switchboard.
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Figure 192: Queue Agent
Global Queue Settings
As explained before, under this section users can configure the feature codes for Dynamic agent login and logout, and also can now customize the keys for virtual queue options like shown below.
Figure 193: Global Queue Settings
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Table 91: Global Queue Settings
Dynamic Agent Login Settings
Agent Login Code Suffix
Configure the code to dial after the queue extension to log into the queue (i.e. queue extension + suffix). If no suffix is configured, dynamic agents will not be able to log in
Agent Logout Code Suffix
Configure the code to dial after the queue extension to log out of the queue (i.e. queue extension + suffix). If no suffix is configured, dynamic agents will not be able to log out.
Virtual Queue Callback Key Settings
Call Back Current Number
Press the feature key configured to set your current number as callback number.
Custom Callback Number
Press these feature key configured to set a custom callback number.
Continue Waiting
Press the feature key configured to continue waiting.
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PICKUP GROUPS
The UCM630X supports pickup group feature which allows users to pick up incoming calls for other extensions if they are in the same pickup group, by dialing "Pickup Extension" feature code (by default *8).
Configure Pickup Groups
Pickup groups can be configured via Web GUICall FeaturesPickup Groups.
Click on Click on
to create a new pickup group. to edit the pickup group.
Click on to delete the pickup group. Select extensions from the list on the left side to the right side.
Figure 194: Edit Pickup Group
Configure Pickup Feature Code
When picking up the call for the pickup group member, the user only needs to dial the pickup feature code. It is not necessary to add the extension number after the pickup feature code. The pickup feature code is configurable under Web GUICall FeaturesFeature Codes. The default feature code for call pickup extension is *8, otherwise if the person intending to pick up the call knows the ringing extension they can use ** followed by the extension number in order to perform the call pickup operation. The following figure shows where you can customize these features codes
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Figure 195: Edit Pickup Feature Code
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MUSIC ON HOLD
Music On Hold settings can be accessed via Web GUIPBX SettingsMusic On Hold. In this page, users could configure music on hold class and upload music files. The "default" Music On Hold class already has 5 audio files defined for users to use.
Figure 196: Music On Hold Default Class
Click on "Create New MOH Class" to add a new Music On Hold class.
Click on Click on
to configure the MOH class sort method to be "Alpha" or "Random" for the sound files. next to the selected Music On Hold class to delete this Music On Hold class.
Click on
to start uploading. Users can upload:
Single files with 8KHz Mono Music file, or Music on hold files in a compressed package with .tar, .tar.gz and .tgz as the suffix. The file name can
only be letters, digits, or special characters -_ the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire
MoH. Users could also download all the music on hold files from UCM. In the Music On Hold page, click on
and the file will be downloaded to your local PC.
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Click on Click on
to disable it from the selected Music On Hold Class. to enable it from the selected Music On Hold Class.
Select the sound files and click on
to delete all selected Music On Hold files.
The UCM630X allows Users to select the Music On Hold file from WebGUI to play it. The UCM630X will initiate a call to the selected extension and play this Music On Hold file once the call is answered.
Steps to play the Music On Hold file: 1. Click on the button for the Music On Hold file.
2. In the prompted window, select the extension to playback and click
.
Figure 197: Play Custom Prompt
3. The selected extension will ring. 4. Answer the call to listen to the music playback.
Users could also record their own Music On Hold to override an existing custom prompt, this can be done by following those steps:
1. Click on . 2. A message of confirmation will pop up, as shown below.
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3. Click
.
Figure 198: Information Prompt
4. In the prompted window, select the extension to playback and click
.
Figure 199: Record Custom Prompt
5. Answer the call and start to record your new music on hold. 6. Hangup the call and refresh Music On Hold page then you can listen to the new recorded file.
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Notes: Once the MOH file is deleted, there are two ways to recover the music files. Users could download the MOH file from this link:
http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-moh-opsound-wav-2.03.tar.gz After downloading and unzip the pack, users could then upload the music files to UCM. Factory reset could also recover the MOH file on the UCM. --------------------------------------------------------------------------------------------------------------------------------------------------
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BUSY CAMP-ON
The UCM630X supports busy camp-on/call completion feature that allows the PBX to camp on a called party and inform the caller as soon as the called party becomes available given the previous attempted call has failed.
The configuration and instructions on how to use busy camp-on/call completion feature can be found in the following guide:
http://www.grandstream.com/sites/default/files/Resources/ucm6xxx_busy_camp_on_guide.pdf
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PRESENCE
UCM does support SIP presence feature which allows users to advertise their current availability status and willingness to receive calls, this way other users can use their phones in order to monitor the presence status of each user and decide whether to call them or not based on their advertised availability.
This feature is different than BLF which is used to monitor the dialog status for each extension (Ringing, Idle or Busy). Instead the SIP presence module gives more options for users to choose which state they want to put themselves in.
In order to configure the presence status of an extension from the web GUI, users can access the menu of configuration using one of the two following methods:
From admin account, go under the menu Extension/TrunkExtensions and choose the desired extension to edit then navigate to the "Features" tab.
OR
From the User Portal, go under the menu Basic InformationExtensions and navigate to the Features tab to have the following options.
Figure 200: SIP Presence Configuration
Select which status to set from the presence status selection drop list, six options are available and below is a brief description of these states:
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Available Away Chat Do Not Disturb Custom Presence Status Unavailable
Table 92: SIP Presence Status
The contact is online and can participate in conversations/phone calls. The contact is currently away (ex: for lunch break). The contact has limited conversation flexibility and can only be reached via chat. The Contact is on DND (Do Not Disturb) mode. Please enter the presence status for this mode on the Web GUI. Up to 64 characters. The contact is unreachable for the moment, please try to contact later.
Another option to set the presence status and which is more practical is using the feature code from the user's phone, one the user dials the feature code (default is *48), a prompt will be played to select which status they want to put themselves in, by pressing the corresponding key.
The feature code can be enabled and customized from the Web GUICall FeaturesFeature Codes.
Figure 201: SIP Presence Feature Code
When a user does change his/her SIP presence status by making a call using presence feature code, the UCM will create a corresponding CDR entry showing the call as Action type = PRSENCE_STATUS.
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Figure 202: Presence Status CDR
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FOLLOW ME
Follow Me is a feature on the UCM630X that allows users to direct calls to other phone numbers and have them ring all at once or one after the other. Calls can be directed to users' home phone, office phone, mobile and etc. The calls will get to the user no matter where they are. Follow Me option can be found under extension settings page Web GUIExtension/TrunkExtensions.
To configure follow me:
1. Choose the extension and click on . 2. Go to the Follow me tab to add destination numbers and enable the feature.
Figure 203: Edit Follow Me
3. Click on
to add local extensions or external numbers to be called after ringing the extension
selected in the first step.
4. Once created, it will be displayed on the follow me list. And you can click on to delete the Follow Me.
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The following table shows the Follow Me configuration parameters:
Enable Skip Trunk Auth Music On Hold Class Confirm When Answering Enable Destination
Default Destination
Follow Me Numbers New Follow Me Number Dialing Order
Table 93: Follow Me Settings
Configure to enable or disable Follow Me for this user.
If external number is added in the Follow Me, please make sure this option is enabled or the "Skip Trunk Auth" option of the extension is enabled, otherwise the external Follow Me number cannot be reached.
Configure the Music On Hold class that the caller would hear while tracking users
By default, it is enabled, and user will be asked to press 1 to accept the call or to press 2 to reject the call after answering a Follow Me call. If it is disabled, the Follow Me call will be established once after the user answers.
When enabled, the call will be routed to the default destination if no one in the Follow Me extensions answers the call.
Configure the destination if no one in the Follow Me extensions answers the call. The available options are: Extension Voicemail Queues Ring Group Voicemail Group IVR External Number
The added numbers are listed here. Click on
to arrange the order. Click
on to delete the number. Click on
to add new numbers.
Add a new Follow Me number which could be a `Local Extension' or `External
Number'. The selected dial plan should have permissions to dial the defined
external number.
Select the order in which the Follow Me destinations will be dialed to reach the user: ring all at once or ring one after the other.
Click on "Follow Me Options" under Web GUIExtension/TrunkExtension page to enable or disable the options listed in the following table.
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Playback Incoming Status Message Record the Caller's Name Playback Unreachable Status Message
Table 94: Follow Me Options
If enabled, the PBX will playback the incoming status message before starting the Follow Me steps. If enabled, the PBX will record the caller's name from the phone so it can be announced to the callee in each step. If enabled, the PBX will playback the unreachable status message to the caller if the callee cannot be reached.
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SPEED DIAL
The UCM630X supports Speed Dial feature that allows users to call a certain destination by pressing one or four digits on the keypad. This creates a system-wide speed dial access for all the extensions on the UCM630X.
To enable Speed Dial, on the UCM630X Web GUI, go to page Web GUICall FeaturesSpeed Dial.
User should first click on
. Then decide from one digit up to four digits combination used for Speed
Dial and select a dial destination from "Default Destination". The supported destinations include extension, voicemail, conference room, voicemail group, IVR, ring group, call queue, page group, DISA, Dial by Name and external number.
Figure 204: Speed Dial Destinations
Figure 205: List of Speed Dial
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DISA
In many situations, the user will find the need to access his own IP PBX resources, but he is not physically near one of his extensions. However, he does have access to his own cell phone. In this case, we can use what is commonly known as DISA (Direct Inward System Access). Under this scenario, the user will be able to call from the outside, whether it is using his cell phone, pay phone, regular PSTN, etc. After calling into UCM630X, the user can then dial out via the SIP trunk or PSTN trunk connected to UCM630X as it is an internal extension.
The UCM630X supports DISA to be used in IVR or inbound route. Before using it, create new DISA under Web GUICall FeaturesDISA.
Click on Click on Click on
to add a new DISA. to edit the DISA configuration. to delete the DISA.
Figure 206: Create New DISA
The following table details the parameters to set and configure DISA feature on UCM630X PBX.
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Name Password
Permission
Response Timeout Digit Timeout Allow Hangup Replace Display Name
Table 95: DISA Settings
Configure DISA name to identify the DISA. Configure the password (digit only) required for the user to enter before using DISA to dial out.
Note: The password must be at least 4 digits. Configure the permission level for DISA. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal". If the user tries to dial outbound calls after dialing into the DISA, the UCM630X will compared the DISA's permission level with the outbound route's privilege level. If the DISA's permission level is higher than (or equal to) the outbound route's privilege level, the call will be allowed to go through. Configure the maximum amount of time the UCM630X will wait before hanging up if the user dials an incomplete or invalid number. The default setting is 10 seconds. Configure the maximum amount of time permitted between digits when the user is typing the extension. The default setting is 5 seconds. If enabled, during an active call, users can enter the UCM630X Hangup feature code (by default it is *0) to disconnect the call or hang up directly. A new dial tone will be heard shortly for the user to make a new call. The default setting is "No".
If enabled, the UCM will replace the caller display name with the DISA name.
Once successfully created, users can configure the inbound route destination as "DISA" or IVR key event as "DISA". When dialing into DISA, users will be prompted with password first. After entering the correct password, a second dial tone will be heard for the users to dial out.
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EMERGENCY
UCM supports configuration and management of numbers to be called in emergency situation, thus bypassing the regular outbound call routing process and allowing users in critical situation to dial out for emergency help with the possibility to have redundant trunks as point of exit in case one of the lines is down.
UCM6xxx series are also now in full compliance with Kari's Law and Ray Baum's Act, for more information, please refer to the following links:
https://www.fcc.gov/mlts-911-requirements
http://www.grandstream.com/sites/default/files/Resources/UCM_Emergency_Calls_Guide.pdf
In addition, Emergency calls can be automatically recorded by toggling on the new Auto Record and recordings can be viewed in the new Emergency Recordings tab on the same page. Additionally, users can have these recordings be sent to the configured email address(es).
Email alerts are also supported after enabling the notification for the event under "Maintenance System Events"
To configure emergency numbers, users need to follow below steps:
1. Navigate on the web GUI under "Call Features Emergency Calls"
2. Click on
to add a new emergency number.
3. Configure the required fields "Name, Emergency Number and Trunk(s) to be used to reach the number". 4. Save and apply the configuration.
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Figure 207: Emergency Number Configuration
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The table below gives more description of the configuration Parameters when creating emergency numbers.
Name Emergency Number Emergency Level
Disable Hunt on Busy
Custom Prompt
Use Trunks Members Notified Strip Prepend Auto Record Send Recording File Email Address
Table 96: Emergency Numbers Parameters
Configure the name of the emergency call. For example, "emergency911","emergency211" and etc. Config the emergency service number. For example,"911","211" and etc. Select the emergency level of the number. Level "3" means the most urgent. If this option is not enabled, when the lines of trunks which the coming emergency call routes by are completely occupied, the line-grabbing function will automatically cut off a line from all busy lines so that the coming emergency call can seize it for dialing out. This option is not enabled by default. This option sets a custom prompt to be used as an announcement to the person receiving an emergency call. The file can be uploaded from the page "Custom Prompt". Click "Prompt" to add additional record. Select the trunks for the emergency call. Select one trunk at least and select five trunks at most. Select the members who will be notified when an emergency call occurs. Specify the number of digits that will be Stripped from the beginning of the dialed number before the call is placed via the selected trunk. Specify the digits to be Prepended before the call is placed via the trunk. Those digits will be prepended after the dialing number is stripped. When enabled, emergency call will be automatically recorded.
When enabled recording files will be sent to the configured email address.
The email address to where the recording files will be sent.
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Figure 208: 911 Emergency Sample
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CALLBACK
Callback is designed for users who often use their mobile phones to make long distance or international calls which may have high service charges. The callback feature provides an economic solution for reduce the cost from this.
The callback feature works as follows:
1. Configure a new callback on the UCM630X. 2. On the UCM630X, configure destination of the inbound route for analog trunk to callback. 3. Save and apply the settings. 4. The user calls the PSTN number of the UCM630X using the mobile phone, which goes to callback
destination as specified in the inbound route. 5. Once the user hears the ringback tone from the mobile phone, hang up the call on the mobile phone. 6. The UCM630X will call back the user. 7. The user answers the call. 8. The call will be sent to DISA or IVR which directs the user to dial the destination number. 9. The user will be connected to the destination number.
In this way, the calls are placed and connected through trunks on the UCM630X instead of to the mobile phone directly. Therefore, the user will not be charged on mobile phone services for long distance or international calls.
To configure callback on the UCM630X, go to Web GUICall FeaturesCallback page and click on
. Configuration parameters are listed in the following table.
Name CallerID Pattern
Outbound Prepend Delay Before Callback Destination
Table 97: Callback Configuration Parameters
Configure a name to identify the Callback. (Enter at least two characters) Configure the pattern of the callers allowed to use this callback. The caller who places the inbound call needs to have the CallerID match this pattern so that the caller can get callback after hanging up the call. Note: If leaving as blank, all numbers are allowed to use this callback. Configure the prepend digits to be added at before dialing the outside number. The number with prepended digits will be used to match the outbound route. `-' is the connection character which will be ignored. Configure the number of seconds to be delayed before calling back the user. Configure the destination which the callback will direct the caller to. Two destinations are available: IVR DISA The caller can then enter the desired number to dial out via UCM630X trunk.
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BLF AND EVENT LIST
BLF
The UCM630X supports BLF monitoring for extensions, ring group, call queue, conference room and parking lot. For example, on the user's phone, configure the parking lot number 701 as the BLF monitored number. When there is a parked call on 701, the LED for this BLF key will light up in red, meaning a call is parked against this parking lot. Pressing this BLF key can pick up the call from this parking lot.
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Note:
On the Grandstream GXP series phones, the MPK supports "Call Park" mode, which can be used to park the call by configuring the MPK number as call park feature code (e.g., 700). MPK "Call Park" mode can also be used to monitor and pickup parked call if the MPK number is configured as parking lot (e.g., 701). ---------------------------------------------------------------------------------------------------------------------------------------------------
Event List
Besides BLF, users can also configure the phones to monitor event list. In this way, both local extensions on the same UCM630X and remote extensions on the VOIP trunk can be monitored. The event list setting is under Web GUICall FeaturesEvent List. Click on "Add" to add a new event list. Sort selected extensions manually in the Eventlist Click on to edit the event list configuration.
Click on to delete the event list.
URI Local Extensions Remote Extensions Special Extensions
Table 98: Event List Settings
Configure the name of this event list (for example, office_event_list). Please note the URI name cannot be the same as the extension name on the UCM630X. The valid characters are letters, digits, _ and -. Select the available extensions/Extension Groups listed on the local UCM630X to be monitored in the event list. If LDAP sync is enabled between the UCM630X and the peer UCM630X, the remote extensions will be listed under "Available Extensions". If not, manually enter the remote extensions under "Special Extensions" field. Manually enter the remote extensions in the peer/register trunk to be monitored in the event list. Valid format: 5000,5001,9000
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Figure 209: Create New Event List
Remote extension monitoring works on the UCM630X via event list BLF, among Peer SIP trunks or Register SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the UCM630X first before using remote BLF feature. Please note the SIP end points need support event list BLF in order to monitor remote extensions.
When an event list is created on the UCM630X and remote extensions are added to the list, the UCM630X will send out SIP SUBSCRIBE to the remote UCM630X to obtain the remote extension status. When the SIP end points register and subscribe to the local UCM630X event list, it can obtain the remote extension status from this event list. Once successfully configured, the event list page will show the status of total extension and
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subscribers for each event list. Users can also select the event URI to check the monitored extension's status and the subscribers' details.
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Notes: To configure LDAP sync, please go to UCM630X Web GUIExtension/TrunkVoIP Trunk. You will see
"Sync LDAP Enable" option. Once enabled, please configure password information for the remote peer UCM630X to connect to the local UCM630X. Additional information such as port number, LDAP outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM630X and remote UCM630X need enable LDAP sync option with the same password for successful connection and synchronization. Currently LDAP sync feature only works between two UCM630Xs. (Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is non-UCM630X PBX. However, it might not work the other way around depending on whether the non-UCM630X PBX supports event list BLF or remote monitoring feature. ---------------------------------------------------------------------------------------------------------------------------------------------------
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DIAL BY NAME
Dial by Name is a feature on the PBX that allows caller to search a person by first or last name via his/her phone's keypad. The administrator can define the Dial by Name directory including the desired extensions in the directory and the searching type by "first name" or "last name". After dialing in, the PBX IVR/Auto Attendant will guide the caller to spell the digits to find the person in the Dial by Name directory. This feature allows customers/clients to use the guided automatic system to contact the enterprise employees without having to know the extension number, which brings convenience and improves business image for the enterprise.
Dial by Name Configuration
The administrators can create the dial by name group under Web GUICall FeaturesDial By Name.
Figure 210: Create Dial by Name Group
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Figure 211: Configure Extension First Name and Last Name
1. Name Enter a Name to identify the Dial by Name group.
2. Extension Configure the direct dial extension for the Dial By Name group.
3. Custom Prompt This option sets a custom prompt for directory to announce to a caller. The file can be uploaded from the page "Custom Prompt". Click "Upload Audio File" to add additional record.
4. Available Extensions/Selected Extensions Select available extensions from the left side to the right side as the directory for the Dial By Name group. Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group. The extensions here must have a valid first name and last name configured under Web GUIExtension/TrunkExtensions in order to be searchable in Dial By Name directory through IVR. By specifying the extensions here, the administrators can make sure unscreened calls will not reach the company employee if he/she does not want to receive them directly.
5. Prompt Wait Time Configure "Prompt Wait Time" for Dial By Name feature. During Dial By Name call, the caller will need to input the first letters of First/Last name before this wait time is reached. Otherwise, timeout will occur, and the call might hang up. The timeout range is between 3 and 60 seconds.
6. Query Type Specify the query type. This defines how the caller will need to enter to search the directory. By First Name: enter the first 3 digits of the first name to search the directory. By Last Name: enter the first 3 digits of the last name to search the directory.
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7. Select Type Specify the select type on the searching result. The IVR will confirm the name/number for the party the caller would like to reach before dialing out. By Order: After the caller enters the digits, the IVR will announce the first matching party's name and number. The caller can confirm and dial out if it is the destination party, or press * to listen to the next matching result if it is not the desired party to call. By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can press number 1 to 8 to select and call or press 9 for results in next page.
The Dial by Name group can be used as the destination for inbound route and key pressing event for IVR. The group name defined here will show up in the destination list when configuring IVR and inbound route. If Dial by Name is set as a key pressing event for IVR, user could use `*' to exit from Dial by Name, then re-enter IVR and start a new event. The following example shows how to use this option.
Figure 212: Dial By Name Group In IVR Key Pressing Events
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Figure 213: Dial By Name Group In Inbound Rule
Please refer to [Username Prompt Customization] for User Name Prompt Customization.
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ACTIVE CALLS AND MONITOR
The active calls on the UCM630X are displayed in Web GUISystem StatusActive Calls page. Users can monitor the status, hang up the call as well as barge in the active calls in real time manner.
Active Calls Status
To view the status of active calls, navigate to Web GUISystem StatusActive Calls. The following figure shows extension 1004 is calling 1000. 1000 is ringing.
Figure 214: StatusPBX StatusActive Calls - Ringing
The following figure shows the call between 1000 and 5555 is established.
Figure 215: StatusPBX StatusActive Calls Call Established
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The gray color of the active call means the connection of call time is less than half an hour. It means this call is normal.
Figure 216: Call Connection less than half hour
The orange color of the active call means the connection of call time is greater than half an hour but less than one hour. It means this call is a bit long.
Figure 217: Call Connection between half an hour and one hour
The red color of the active call means the connection of call time is more than one hour. It means this call could be abnormal.
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Hang Up Active Calls
Figure 218: Call Connection more than one hour
To hang up an active call, click on to hang up all active calls.
Call Monitor
During an active call, click on icon
icon in the active call dialog. Users can also click on and the monitor dialog will pop up.
Figure 219: Configure to Monitor an Active Call
In the "Monitor" dialog, configure the following to monitor an active call: 1. Enter an available extension for "Monitor's Extension" which will be used to monitor the active call.
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2. "Monitored Extension" must be one of the parties in the active call to be monitored.
3. Select spy mode. There are three options in "Spy Mode".
Listen In "Listen" mode, the extension monitoring the call can hear both parties in the active call but the audio of the user on this extension will not be heard by either party in the monitored active call.
Whisper In "Whisper" mode, the extension monitoring the call can hear both parties in the active call. The user on this extension can only talk to the selected monitored extension and he/she will not be heard by the other party in the active call. This can be usually used to supervise calls.
Barge In "Barge" mode, the extension monitoring the call can talk to both parties in the active call. The call will be established similar to three-way conference.
4. Enable or disable "Require Confirmation" option. If enabled, the confirmation of the invited monitor's extension is required before the active call can be monitored. This option can be used to avoid adding participant who has auto-answer configured, or call forwarded to voicemail.
5. Click on "Add". An INVITE will be sent to the monitor's extension. The monitor can answer the call and start monitoring. If "Require Confirmation" is enabled, the user will be asked to confirm to monitor the call.
Another way to monitor active calls is to dial the corresponding feature codes from an extension. Please refer to [Table 99: UCM630X Feature Codes] and [Call Recording] section for instructions.
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CALL FEATURES
The UCM630X supports call recording, transfer, call forward, call park and other call features via feature code. This section lists all the feature codes in the UCM630X and describes how to use the call features.
Feature Codes
Feature Maps Blind Transfer
Attended Transfer Seamless Transfer
Table 99: UCM630X Feature Codes
Default code: #1 Enter the code during active call. After hearing "Transfer", you
will hear dial tone. Enter the number to transfer to. Then the user will be disconnected, and transfer is completed. Options: Disable Allow Caller: Enable the feature code on caller side only. Allow Callee: Enable the feature code on callee side only. Allow Both: Enable the feature code on both caller and callee.
Default code: *2 Enter the code during active call. After hearing "Transfer", you
will hear the dial tone. Enter the number to transfer to and the user will be connected to this number. Hang up the call to complete the attended transfer. In case of the called party does not answer, users could press *0 to cancel the call and retrieve the first call leg. Options: Disable Allow Caller: Enable the feature code on caller side only. Allow Callee: Enable the feature code on callee side only. Allow Both: Enable the feature code on both caller and callee. Default code: *44 (Disabled by default). Seamless Transfer allows user to perform blind transfer using UCM feature code without having music on hold presented during the transfer process, it minimizes the interruption during transfer, making the process smooth and simple. During an active call use the feature code (*44 by default) followed by the number you want to transfer to in order to perform the seamless transfer.
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Disconnect Call Park Start/Stop Call Recording
Default code: *0 Enter the code during active call. It will disconnect the call. Options:
Disable Allow Caller: Enable the feature code on caller side only. Allow Callee: Enable the feature code on callee side only. Allow Both: Enable the feature code on both caller and callee.
Default code: #72 Enter the code during active call to park the call. Options:
Disable Allow Caller: Enable the feature code on caller side only. Allow Callee: Enable the feature code on callee side only. Allow Both: Enable the feature code on both caller and callee.
Default code: *3 Enter the code followed by # or SEND to start recording the
audio call and the UCM630X will mix the streams natively on the fly as the call is in progress. Options: Disable Allow Caller: Enable the feature code on caller side only. Allow Callee: Enable the feature code on callee side only. Allow Both: Enable the feature code on both caller and callee.
Enable Recording Whitelist
Enable the Recording Whitelist feature
Recording Operation Whitelist Feature Code Digits Timeout DND/Call Forward
Select extension in the whitelist that can use the *3 recording function.
Set the maximum interval (ms) between digits for feature code activation
Do Not Disturb (DND) Activate Default code: *77
Do Not Disturb (DND) Deactivate Default code: *78
Call Forward Busy Activate
Default Code: *90 Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.
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Call Forward Busy Deactivate
Default Code: *91
Default Code: *92 Call Forward No Answer Activate Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.
Call Forward No Answer Deactivate
Default Code: *93
Call Forward Unconditional Activate Call Forward Unconditional Deactivate
Remote Call Forward Enable
Feature Codes
Voicemail Access Code
Default Code: *72 Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.
Default Code: *73
Enable this option and configure the Remote Call Forward Whitelist below to allow specific extensions to dial the remote call forwarding feature codes to set call forwarding for any extension.
Default Code: *98 Enter *98 and follow the voice prompt. Or dial *98 followed by
the extension and # to access the entered extension's voicemail box.
My Voicemai Agent Pause Agent Unpause
Default Code: *97 Press *97 to access the voicemail box.
Default Code: *83 Pause the agent in all call queues.
Default Code: *84 Unpause the agent in all call queues.
Paging Prefix
Default Code: *81 To page an extension, enter the code followed by the extension
number.
Intercom Prefix Blacklist Add
Default Code: *80 To intercom an extension, enter the code followed by the
extension number.
Default Code: *40 To add a number to blacklist for inbound route, dial *40 and
follow the voice prompt to enter the number.
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Blacklist Remove
Default Code: *41 To remove a number from current blacklist for inbound route,
dial *41 and follow the voice prompt to remove the number.
Call Pickup on Ringing Pickup In-call Pickup Extension
Default Code: ** To pick up a call for any extension xxxx, enter the code followed
by the extension number xxxx.
Default Code: *45 (Disabled by default). If "Pickup In-call" feature is enabled, only the extensions added
in "Allowed to seamless transfer" in the extension's Seamless Transfer Privilege Control List" can pick up the call.
Default Code: *8 This code is for the pickup group, which can be assigned for
each extension on the extension configuration page. If there is an incoming call to an extension, the other extensions
within the same pickup group can dial *8 directly to pick up the call.
Direct Dial Voicemail Prefix
Direct Dial Mobile Phone Prefix Call Completion Request Call Completion Cancel Enable Spy
Default Code: * This code is for the user to directly dial or transfer to an
extension's voicemail. For example, directly dial *5000 will have to call go into the
extension 5000's voicemail. If the user would like to transfer the call to the extension 5000's voicemail, enter *5000 as the transfer target number.
Default Code: *88 If you have the permission to call mobile phone number, use
this prefix plus the extension number can dial the mobile phone number of this extension directly. Default Code: *11 This code is for the user who wants to use Call Completion to complete a call. Default Code: *12 This code is for the user who wants to cancel Call Completion request.
Check this box to enable spy feature codes. Disabled by default.
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Listen Spy Whisper Spy Barge Spy Wakeup Service PMS Wakeup Service PMS Remote Wakeup Service Update PMS Room Status
Presence Status
Dynamic Agent Logout
Default Code: *54 ("Enable Spy" needs to be checked) This is the feature code to listen in on a call to monitor
performance. Monitor's line will be muted, and neither party will hear from the monitor's extension.
Default Code: *55 ("Enable Spy" needs to be checked) This is the feature code to speak to one side of the call (for
example, whisper to employees to help them handle a call). Only one side will be able to hear from the monitor's extension.
Default Code: *56 ("Enable Spy" needs to be checked) This is the feature code to join in on the call to assist both
parties.
Default Code: *36 Dial this code to access UCM wakeup service, you can add,
update, activate or deactivate wakeup service.
Default Code: *35 Dial this code to access UCM PMS wakeup service, you can
add, update, activate or deactivate PMS wakeup service.
Default Code: *37 Allows the user to add, update, activate, and deactivate PMS
wakeup service for other extensions.
Default Code: *23 Use this code with maid code to update PMS room status.
Choose the status to set after hearing the prompt, for example: for maid 001 dial *23001 and then 1 after hearing the prompt.
Dial this code to set the presence status of the extension. Possible options are:
1:"unavailable" 2:"available" 3:"away" 4:"chat" 5:"dnd" 6:"userdef"
Default Code: *85 Use this code to logout the dynamic agent from all queues.
The UCM630X also allows user to one click enable / disable specific feature code as shown below:
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Figure 220: Enable/Disable Feature codes
Parking Lot
User can create parking lots and their related slots under Web GUI Call Features Parking Lot. In the Parking Lot page, users can create lots of their own. This allows different groups within an organization to have their own parking lots instead of sharing one large parking lot with others. While creating a new parking lot, users can assign it a range that they think is appropriate for the group that will use the parking lot.
Figure 221: Parking Lot
User can create a new Parking lot by clicking on button "Add" :
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Parking Lot Extension Parking Lot Name Parked Slots
Use Parklot as Extension
Parking Timeout (s) Music On Hold Classes Failover Destination
Ring All Callback on Timeout Forward to destination on timeout
Figure 222: New Parking Lot
Table 100 : Parking Lot
Default Extension: 700 During an active call, initiate blind transfer and then enter this
code to park the call. Set a name to the parking lot Default Extension: 701-720 These are the extensions where the calls will be parked, i.e.,
parking lots that the parked calls can be retrieved.
If checked, the parking lot number can be used as extension. The user can transfer the call to the parking lot number to park the call. Please note this parking lot number range might conflict with extension range.
Default setting is 300 seconds, and the maximum limit is 99.999 seconds.
This is the timeout allowed for a call to be parked. After the timeout, if the call is not picked up, the extension who parks the call will be called back.
Select the Music on Hold Class.
Configures a callback failover destination when the extension that is called back is busy. The call will be routed to the destination number and this reduces the chance of dropping parked calls.
If enabled, all registered endpoints of the extension will ring when callback occurs. Otherwise, only the original endpoint will be called back. If enabled, the call will be routed to the configured destination upon timeout. Otherwise, the call will be routed back to the original caller.
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Timeout Destination Parking Lot Timeout Alert-Info
This option appears once Forward to Destination on Timeout is enabled. Upon park timeout, the call will be routed to the configured destination.
Adds an Alert-Info header to parking lot callbacks after the Parking Timeout has been reached.
Call Park
The UCM630X provides call park and call pickup features via feature code.
Park a Call
There are two feature codes that can be used to park the call.
Feature MapsCall Park (Default code #72) During an active call, press #72 and the call will be parked. Parking lot number (default range 701 to 720) will be announced after parking the call.
Feature MiscCall Park (Default code 700) During an active call, initiate blind transfer (default code #1) and then dial 700 to park the call. Parking lot number (default range 701 to 720) will be announced after parking the call.
Retrieve Parked Call
To retrieve the parked call, simply dial the parking lot number and the call will be established. If a parked call is not retrieved after the timeout, the original extension who parks the call will be called back.
Monitor Call Park CID Name Information (GXP21xx, GRP261x Phones Only)
Users can see the CID name information of parked calls. VPK/MPKs must be configured as "Monitored Call Park" with the desired parking lot extension. The display will alternate between displaying the parking lot extension and the call's CID name. There is no need to configure anything on the UCM.
.
Call Recording
Figure 223: Monitored Call Park CID name
The UCM630X allows users to record audio during the call. If "Auto Record" is turned on for an extension, ring group, call queue or trunk, the call will be automatically recorded when there is established call with it. Otherwise, please follow the instructions below to manually record the call.
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1. Make sure the feature code for "Start/Stop Call Recording" is configured and enabled. 2. After establishing the call, enter the "Start/Stop Call Recording" feature code (by default it is *3) followed by
# or SEND to start recording. 3. To stop the recording, enter the "Start/Stop Call Recording" feature code (by default it is *3) followed by # or
SEND again. Or the recording will be stopped once the call hangs up.
4. The recording file can be retrieved under Web GUICDR. Click on click on to download the recording file.
to show and play the recording or
Figure 224: Download Recording File from CDR Page
The above recorded call's recording files are also listed under the UCM630X Web GUICDRRecording Files.
Figure 225: Download Recording File from Recording Files Page
Enable Spy
If "Enable Spy" option is enabled, feature codes for Listen Spy, Whisper Spy and Barge Spy are available for users to dial from any extension to perform the corresponding actions. Assume a call is on-going between extension A and extension B, user could dial the feature code from extension C to listen on their call (*54 by default), whisper to one side (*55 by default), or barge into the call (*56 by default). Then the user will be asked to enter the number to call, which should be either side of the active call, extension A or B in this example.
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---------------------------------------------------------------------------------------------------------------------------------------------------
Caution: "Enable Spy" allows any user to listen to any call by feature codes. This may result in the leakage of user privacy. ---------------------------------------------------------------------------------------------------------------------------------------------------
Shared Call Appearance (SCA)
Shared Call Appearance (SCA) functionality has been added to the UCM. With SCA, users can assign multiple devices to one extension, configure endpoints to monitor that extension, make actions on behalf of that extension such as viewing call status and placing and receiving calls, and even barging into existing calls. To configure the SCA functionality, please follow the steps below:
1. Users can enable SCA by navigating to the Extensions page, editing the desired extension, and enabling the option SCA. Note: With SCA enabled, the Concurrent Registrations field can only have a value of 1.
Figure 226: Enabling SCA option under Extension's Settings
2. After enabling the option, navigate to Call FeaturesSCA. The newly enabled SCA extension will be listed. Click the "+" button under the Options column to add a number that will share the main extension's call appearance, which will be called private numbers.
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Figure 227: SCA Number Configuration
3. Configure the private number as desired.
Figure 228: SCA Private Number Configuration
4. Once the private number has been created, users must now register a device to it. To properly register a device to the private number, use the configured private number as the SIP User ID. Auth ID and Password will be the same as the main extensions. Once registration is complete, SCA is now configured.
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Figure 229: SCA Options
5. Next, configure the VPK or MPK to Shared for both the main extension and the private number. SCA is now configured for both endpoint devices.
The following table describe the SCA Number configuration setting:
Private Number Related Shared Line Enable This Number
Allow Origination from This Number
Table 101: Add SCA Private Number
Configures the private number for the SCA.
Display the related shared line. Whether enable this private number. If not enabled, this private number is only record in DB, it will not affect other system feature. Enable this option will allow calling from this private number. By default, it is enabled.
Allow Termination to This Number
Enable this option will allows calls to this private number. By default, it is enabled.
The following table describes the options available when editing the SCA number:
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Table 102: Editing the SCA Number
Shared Line Number
While SCA is enabled, this number will be the same as the extension number.
Allow Call Retrieve from Another Allows remote call retrieval. Must be enabled in public hold. By
Location
default, it is enabled.
Alert All Appearances for Group Paging Calls
Allows all SCA group members to ring when the SCA shared number is paged. If disabled, only the SCA shared number will ring when paged. By default, it is disabled.
Multiple Call Arrangement
Allows simultaneous calls in an SCA group. By default, it is disabled.
Allow Bridging between Locations
Bridge Warning Tone
Allows location bridging for SCA group. Must be enabled when using the Barge-In feature. By default, it is disabled.
Configures the notification in the bridge when another party join. None: No notification sound. Barge-In only: Notification sound will play when another party join. Barge-In and Repeat: Notification sound will play when another party joins and repeat every 30 seconds.
By default, it is set to "Barge-In Only".
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ANNOUNCEMENT
The Announcement feature (not to be confused with Announcement Paging and Announcement Center) is a feature that allows users to set an unskippable audio file to play to callers before routing them to a configured destination. Announcements can be configured as a destination in the Inbound Routes page.
To configure Announcement, users need to follow below steps:
1. Navigate on the web GUI under "Call Features Announcement"
2. Click on
to add a new Announcement.
3. Configure the required fields Name, Prompt, Default Destination to be used for the announcement. Save and apply the configuration.
Figure 230: Announcement settings
The table below gives more description of the configuration parameters when creating Announcement.
Name Prompt Default Destination
Table 103: Announcement Parameters
Configure the name of the Announcement. Audio file that needs to be uploaded in order to be played for a specific destination. Select the destination where to play the audio file.
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PBX SETTINGS
This section describes internal options that have not been mentioned in previous sections yet. The settings in this section can be applied globally to the UCM630X, including general configurations, jitter buffer, RTP settings, ports config and STUN monitor. The options can be accessed via Web GUIPBX SettingsGeneral Settings.
PBX Settings/General Settings
Table 104: Internal Options/General
General Preferences
Global Outbound CID
Configure the global CallerID used for all outbound calls when no other CallerID is defined with higher priority. If no CallerID is defined for extension or trunk, the global outbound CID will be used as CallerID.
Global Outbound CID Name
Configure the global CallerID Name used for all outbound calls. If configured, all outbound calls will have the CallerID Name set to this name. If not, the extension's CallerID Name will be used.
Ring Timeout
Configure the number of seconds to ring an extension before the call goes to the user's voicemail box. The default setting is 60. Note: This is the global value used for each extension if "Ring Timeout" field is left empty on the extension configuration page.
Call Duration Limit
Configure the maximum duration of call-blocking.
Maximum Call Duration The maximum call duration (in seconds). The default value 0 means no limit.
(s)
Warning Time (s)
The amount of seconds before the maximum call duration is reached to play the warning tone to the caller.
Warning Repeat Interval The amount of seconds that must pass after the first warning tone before
(s)
another warning tone is played.
Enable 486 to Failover Trunk
Reroutes failed outbound calls that receive a 486 response through the failover trunk to retry the call. If disabled, calls that receive a 486 response will be terminated.
Record Prompt
If enabled, users will hear voice prompt before recording is started or stopped. For example, before recording, the UCM630X will play voice prompt "The call will be recorded". The default setting is "No".
Device Name
The name of the UCM you are using.
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International Call Prefix Conference Max Concurrent Audio
Conference Voice Indicator Sensitivity
Conference Voice Quality Extension Preferences
When this configuration is empty, International Call Prefix can be empty or +. Maximum number of participants that can be heard simultaneously in audio/video conferences. If the number of participants talking at any given point exceeds this value, the audio of the excess participants will not be heard. Configures the sensitivity of the talking indicator in conferences. Setting this higher will make the talking indicator appear more easily for lower volumes of audio. Note: This does not adjust audio input sensitivity itself. Lower volumes of sounds may still be heard even if the talking indicator does not show the source.
Voice quality of audio and video conferences
If enabled, strong password will be enforced for the password created on the UCM630X. The default setting is "No".
Enforce Strong Passwords
Enable Random Password Enable Auto E-mail To User
Strong Password Rules: 1. Password for voicemail, voicemail group, outbound route, DISA, call queue
and conference require non-repetitive and non-sequential digits, with a minimum length of 4 digits. Repetitive digits pattern (such as 0000, 1111, 1234, 2345, and etc.), or common digits pattern (such as 111222, 321321 and etc.) are not allowed to be configured as password.
2. Password for extension registration, Web GUI admin login, LDAP and LDAP sync requires alphanumeric characters containing at least two categories of the following, with a minimum length of 4 characters. Numeric digits Lowercase alphabet characters Uppercase alphabet characters Special characters
If enabled, random password will be generated when the extension is created. The default setting is "Yes". It is recommended to enable it for security purpose. If enabled, UCM630X will send Email notification to user automatically after editing extension settings or adding a new extension.
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Disable Extension Range
Extension Ranges
If set to "Yes", users could disable the extension range preconfigured/configured on the UCM630X. The default setting is "No".
Note: It is recommended to keep the system assignment to avoid inappropriate usage and unnecessary issues. The default extension range assignment is:
User Extensions: 1000-6299 User Extensions is referring to the extensions created under Web GUIExtension/TrunkExtensions page.
Pick Extensions: 4000-4999 This refers to the extensions that can be manually picked from end device when being provisioned by the UCM630X. There are two related options in zero config pageZero Config Settings, "Pick Extension Segment" and "Enable Pick Extension". If "Enable Pick Extension" under zero config settings is selected, the extension list defined in "Pick Extension Segment" will be sent out to the device after receiving the device's request. This "Pick Extension Segment" should be a subset of the "Pick Extensions" range here. This feature is for the GXP series phones that support selecting extension to be provisioned via phone's LCD.
Auto Provision Extensions: 5000-6299 This sets the range for "Zero Config Extension Segment" which is the extensions can be assigned on the UCM630X to provision the end device.
Conference Extensions: 6300-6399 Ring Group Extensions: 6400-6499 Queue Extensions: 6500-6599 Voicemail Group Extensions: 6600-6699 IVR Extensions: 7000-7100 Dial By Name Extensions: 7101-7199
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PBX Settings/RTP Settings RTP Settings
RTP Start RTP End
Table 105: Internal Options/RTP Settings
Configure the RTP port starting number. The default setting is 10000. Configure the RTP port ending address. The default setting is 20000.
Strict RTP
Configure to enable or disable strict RTP protection. If enabled, RTP packets that do not come from the source of the RTP stream will be dropped. The default setting is "Disable".
RTP Checksums ICE Support
STUN Server
Configure to enable or disable RTP Checksums on RTP traffic. The default setting is "Disable".
Configure whether to support ICE. The default setting is enabled. ICE is the integrated use of STUN and TURN structure to provide reliable VoIP or video calls and media transmission, via a SIP request/ response model or multiple candidate endpoints exchanging IP addresses and ports, such as private addresses and TURN server address.
Configure STUN server address. STUN protocol is a Client/Server and also a Request/Response protocol. It is used to check the connectivity between the two terminals, such as maintaining a NAT binding entries keep-alive agreement. The default STUN Server is stun.ipvideotalk.com.
BFCP UDP Start BFCP UDP End BFCP TCP Start BFCP TCP End
TURN Server
TURN Server Name TURN Server Password Connection Protocol
Valid format: [(hostname | IP-address) [':' port] The default port number is 3478 if not specified. Configure BFCP UDP port starting number. The default setting is 50000. Configure BFCP UDP port ending number. The default setting is 52999. Configure BFCP TCP port starting number. The default setting is 53000. Configure BFCP TCP port ending number. The default setting is 55999. Configure TURN server address. TURN is an enhanced version of the STUN protocol and is dedicated to the processing of symmetric NAT problems. Configure turn server account name Configure turn server account password. Protocol used to connect to the TURN server.
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Payload The UCM630X payload type for audio codecs and video codes can be configured here.
AAL2-G.726
DTMF G.721 Compatible
G.726
iLBC OPUS Audio FEC Payload Type Audio RED Payload Type H.264 H.265 H.263P VP8 Main Video FEC RTP FECC RTX G.722.1 G.722.1C
Table 106: Internal Options/Payload
Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword packing). The default setting is 112. Configured payload type for DTMF. The default setting is 101. Configure to enable/disable G.721 compatible. The default setting is Yes. Configure the payload type for G.726 if "G.721 Compatible" is disabled. The default setting is 111. Configure the payload type for iLBC. The default setting is 97. Configure the payload type for OPUS. The default setting is 123.
Configure the Audio FEC Payload Type. The default setting is 127
Configure the Audio RED Payload Type. Default setting is 122
Configure the payload type for H.264. The default setting is 99. Configure the payload type for H.264. The default setting is 114. Configure the payload type for H.263+. The default setting is 100 103. Configure the payload type for VP8. The default setting is 108. Configure the Main Video FEC Configure the RTP FECC Configure the RTX G.722.1Low-complexity coder, 24kbps. G.722.1CLow-complexity coder, 48kbps.
PBX Settings/Voice Prompt Customization Record New Custom Prompt
In the UCM630X Web GUIPBX SettingsVoice PromptCustom Prompt page, click on "Record" and follow the steps below to record new IVR prompt.
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Figure 231: Record New Custom Prompt
1. Specify the IVR file name.
2. Select the format (GSM or WAV) for the IVR prompt file to be recorded.
3. Select the extension to receive the call from the UCM630X to record the IVR prompt. 4. Click the "Record" button. A request will be sent to the UCM630X. The UCM630X will then call the extension
for recording the IVR prompt from the phone.
5. Pick up the call from the extension and start the recording following the voice prompt.
6. The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play, or delete the recording.
Upload Custom Prompt
If the user has a pre-recorded IVR prompt file, click on "Upload" in Web GUIPBX SettingsVoice PromptCustom Prompt page to upload the file to the UCM630X. The following are required for the IVR prompt file to be successfully uploaded and used by the UCM630X: PCM encoded. 16 bits. 8000Hz mono. In .mp3 or .wav format; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix. File size under 5M.
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Figure 232: Upload Custom Prompt
Click on "choose file to upload" to start uploading. Once uploaded, the file will appear in the Custom Prompt web page.
Download All Custom Prompt
On the UCM630X, the users can download all custom prompts from UCM Web GUI to local PC. To download all custom prompt, log in UCM Web GUI and navigate to PBX SettingsVoice PromptCustom Prompt and click on "Download All". The following window will pop up in order to set a name for the downloaded file.
Figure 233: Download All Custom Prompt
Note: The downloaded file will have a .tar extension.
PBX Settings/ Call Failure Tone Settings SIP Trunk Prompt Tone
Prompt Tone Settings tab has been added to the UCM to help users choose which prompt will be played by the UCM during call failure, the following voice message responses have been added and can be set to be played for 4XX, 5XX, and 6XX call failures:
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Default for 404 and 604 status codes: "Your call can't be completed as dialed. Please check the number and dial again."
Default for 5xx status codes: "Server error. Please check your device." Default for 403 and 603 status codes: "The call was rejected by the server. Please try again later." Default for all other status codes: "All circuits are busy now. Please try again later."
Additionally, custom voice messages recorded and uploaded in PBX SettingsVoice PromptCustom Prompt can be used for these failure responses instead of the default messages.
Figure 234: SIP Trunk Prompt Tone
General Call Prompt Tone Moreover, users also have the possibility to customize the prompt for typical call failure reasons like (no permission to allow outbound calls, busy lines, incorrect number dialed ...Etc.).
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To customize these prompts user could record and upload their own files under "'PBX Settings Voice Prompt Custom Prompts" then select each one for specific call failure case under "PBX Settings -> Call Failure Tone Settings General Call Prompt Tone" page as shown on the following figure:
Figure 235: General call Failure Prompts
PBX Settings/Recordings Storage
The UCM630X supports call recordings automatically or manually and the recording files can be saved in external storage plugged in the UCM630X or on the UCM630X locally. To manage the recording storage, users can go to UCM630X Web GUIPBX SettingsRecordings Storage page and select whether to store the recording files in USB Disk, SD card, GDMS or locally on the UCM630X.
Figure 236: SettingsRecordings Storage
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If "Enable Auto Change" is selected, the recording files will be automatically saved in the available USB Disk or SD card plugged into the UCM630X. If both USB Disk and SD card are plugged in, the recording files will be always saved in the USB Disk.
If "Local" is selected, the recordings will be stored in UCM630X internal storage. If "GDMS Cloud Storage" is selected, recording data will no longer be stored locally and if you need to listen
to the recording, download the recording file to the computer side and play it offline. If "USB Disk" or "SD Card" is selected, the recordings will be stored in the corresponding plugged in external
storage device. Please note the options "USB Disk" and "SD Card" will be displayed only if they are plugged into the UCM630X. Once "USB Disk" or "SD Card" is selected, click on "OK". The user will be prompted to confirm to copy the local files to the external storage device.
Figure 237: Recordings Storage Prompt Information
Click on "OK" to continue. The users will be prompted a new dialog to select the categories for the files to be copied over.
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Figure 238: Recording Storage Category
On the UCM630X, recording files are generated and exist in 3 categories: normal call recording files, conference recording files, and call queue recording files. Therefore, users have the following options when select the categories to copy the files to the external device:
- Recording Files: Copy the normal recording files to the external device. - Conference: Copy the conference recording files to the external device. - Queue: Copy the call queue recording files to the external device. - All: Copy all recording files to the external device.
PBX Settings/NAS
The UCM supports adding and backing up recordings to a network-attached storage (NAS) server. Following table describes NAS settings:
Enable
Host
Share Name Username Password Status
Table 107: NAS Settings
Enabled / Disable the NAS recording functionality. Configure the Domain or IP address of the NAS server. Note: Currently, only IP addresses are supported in the Host/IP field. Specify the name of the shared folder. Specify the account username to access the NAS server. Configure the account password to access the NAS server. If configured correctly, the Status field will show "Mounted", and the newly
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added NAS server will be shown on the Mounted Netdisk List. Additionally, the NAS will appear as a selectable storage option in the PBX SettingsRecording Storage page and CDRRecording Files page.
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SIP SETTINGS
The UCM630X SIP global settings can be accessed via Web GUIPBX SettingsSIP Settings.
SIP Settings/General
Realm For Digest Authentication Bind UDP Port Bind IPv4 Address Bind IPv6 Address
Allow Guest Calls
Allow Transfer
MWI From Enable Diversion Header
Block Collect Calls
Table 108: SIP Settings/General
Configure the host name or domain name for the UCM630X. Realms MUST be globally unique according to RFC3261. The default setting is Grandstream. Configure the UDP port used for SIP. The default setting is 5060. Configure the IPv4 address to bind to. The default setting is 0.0.0.0, which means binding to all addresses. Configure the IPv6 address to bind to. The default is : "[::]" and it means to bind to all IP addresses. If enabled, the UCM630X allows unauthorized INVITE coming into the PBX and the call can be made. The default setting is "No". Warning: Please be aware of the potential security risk when enabling "Allow Guest Calls" as this will allow any user with the UCM630X address to dial into the UCM630X. If set to "No", all transfers initiated by the endpoint in the UCM630X will be disabled (unless enabled in peers or users). The default setting is "Yes". When sending MWI NOTIFY requests, this value will be used in the "From:" header as the "name" field. If no "From User" is configured, the "user" field of the URI in the "From:" header will be filled with this value.
If disabled, the UCM will not forward the diversion header.
If enabled, collect calls will be blocked. Note: Collect calls are indicated by the header "P-Asserted-Service-Info: servicecode=Backward Collect Call, P-Asserted-Service-Info: service-code=Collect Call".
SIP Settings/MISC
Table 109: SIP Settings/Misc
Outbound SIP Registrations
Register Timeout
Configure the register retry timeout (in seconds). The default setting is 20.
Register Attempts
Configure the number of registration attempts before the UCM630X gives up. The
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default setting is 0, which means the UCM630X will keep trying until the server side accepts the registration request.
Video
Max Bit Rate (kb/s)
Configure the maximum bit rate (in kb/s) for video calls. The default setting is 384.
Support SIP Video
Select to enable video support in SIP calls. The default setting is "Yes".
Reject Non-Matching INVITE
If enabled, when rejecting an incoming INVITE or REGISTER request, the UCM630X will always reject with "401 Unauthorized" instead of notifying the requester whether there is a matching user or peer for the request. This reduces the ability of an attacker to scan for valid SIP usernames. The default setting is "No".
SDP Attribute Passthrough
Enable Attribute Passthrough
If enable, and if the service does not know the attribute of FEC/FECC/BFCP, then the attribute will be passthrough.
Early Media
Enable Use Final SDP If enabled, call negotiation will use final response SDP.
Blind Transfer
Allow callback when blind transfer fails
If enabled, the UCM will call back to the transferrer when blind transfer fails (reason of failure includes: busy and no answer). Note: This feature takes effect only on internal calls.
Configure the timeout in (s) for the transferrer waiting for the destination to Blind transfer timeout
answer. Default is 60s.
Hold
Forward HOLD Requests
Configure the UCM to forward HOLD requests instead of processing holds internally. This serves to meet the standards set by some providers that require HOLD requests to be passed along from endpoint to endpoint. This option is disabled by default. Note: Enabling this option may cause hold retrieval issues and MOH to not be heard.
SIP Settings/Session Timer
Force Timer Timer Session Expire
Min SE
Table 110: SIP Settings/Session Timer
If checked, always request, and run session timer. If checked, run session timer only when requested by other UA. Configure the maximum session refresh interval (in seconds). Default is 1800. Configure the minimum session refresh interval (in seconds). The default setting is 90.
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SIP Settings/TCP and TLS
TCP Enable TCP Bind IPv4 Address TCP Bind IPv6 Address TLS Enable
TLS Bind IPv4 Address
TLS Bind IPv6 Address
TLS Do Not Verify
TLS Self-Signed CA
TLS Cert
TLS CA Cert TLS CA List
Table 111: SIP Settings/TCP and TLS
Configure to allow incoming TCP connections with the UCM630X. The default setting is "No". Configure the IP address for TCP server to bind to. "0.0.0.0" means binding to all interfaces. The port number is optional, and the default port number is 5060. For example, 192.168.1.1:5062. Configure the IPv6 address for TCP server to bind to. "[::]" means bind to all interfaces. The port number is optional with the default being 5060. For example, [2001:0DB8:0000:0000:0000:0000:1428:0000]:5060. Configure to allow incoming TLS connections with the UCM630X. The default setting is "Yes".
Configure the IPv4 address for TLS server to bind to. "0.0.0.0" means binding to all interfaces. The port number is optional, and the default port number is 5061. For example, 192.168.1.1:5063. Note: The IP address must match the common name (host name) in the certificate so that the TLS socket will not bind to multiple IP addresses. Configure the IPv6 address for TLS server to bind to. "[::]" means bind to all interfaces. The port number is optional with default being 5061. For example, [2001:0DB8:0000:0000:0000:0000:1428:0000]:5061. Note: The IP address must match the common name (host name) in the certificate so that the TLS socket will not bind to multiple IP addresses. If enabled, the TLS server's certificate will not be verified when acting as a client. The default setting is "Yes".
This is the CA certificate if the TLS server being connected to requires self-signed certificate, including server's public key. This file will be renamed as "TLS.ca" automatically. Note: The size of the uploaded ca file must be under 2MB.
This is the Certificate file (*.pem format only) used for TLS connections. It contains private key for client and signed certificate for the server. This file will be renamed as "TLS.pem" automatically. Note: The size of the uploaded certificate file must be under 2MB. This file must be named with the CA subject name hash value. It contains CA's (Certificate Authority) public key, which is used to verify the accessed servers. Note: The size of the uploaded CA certificate file must be under 2MB. Display a list of files under the CA Cert directory.
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SIP Settings/NAT
External Host
Table 112: SIP Settings/NAT
Configure a static IP address and port (optional) used in outbound SIP messages if the UCM630X is behind NAT. If it is a host name, it will only be looked up once.
If enabled, the SDP connection will use the IP address resolved from the external Use IP address in SDP
host.
External UDP Port
Configure externally mapped UDP port when the PBX is behind a static NAT or PAT.
External TCP Port
Configure the externally mapped TCP port when the UCM630X is behind a static NAT or PAT.
External TLS Port
Configures the externally mapped TLS port when UCM630X is behind a static NAT or PAT.
Local Network Address
Specify a list of network addresses that are considered inside of the NAT network. Multiple entries are allowed. If not configured, the external IP address will not be set correctly. A sample configuration could be as follows: 192.168.0.0/16
SIP Settings/TOS
ToS for SIP ToS for RTP Audio ToS for RTP Video
Table 113: SIP Settings/ToS
Configure the Type of Service for SIP packets. The default setting is None. Configure the Type of Service for RTP audio packets. The default setting is None. Configure the Type of Service for RTP video packets. The default setting is None.
Default Incoming/Outgoing Registration Time
Configure the default duration (in seconds) of incoming/outgoing registration. The default setting is 120.
Max Registration/Subscrip tion Time
Configure the maximum duration (in seconds) of incoming registration and subscription allowed by the UCM630X. The default setting is 3600.
Min Registration/Subscrip tion Time
Configure the minimum duration (in seconds) of incoming registration and subscription allowed by the UCM630X. The default setting is 60.
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Enable Relaxed DTMF DTMF Mode RTP Timeout
Select to enable relaxed DTMF handling. The default setting is "No".
Select DTMF mode to send DTMF. The default setting is RFC4733. If "Info" is selected, SIP INFO message will be used. If "Inband" is selected, a-law or u-law are required. When "Auto" is selected, "RFC4733" will be used if offered, otherwise "Inband" will be used. The default setting is "RFC4733". During an active call, if there is no RTP activity within the timeout (in seconds), the call will be terminated. The default setting is no timeout. Note: This setting does not apply to calls on hold.
RTP Hold Timeout RTP Keep-alive
When the call is on hold, if there is no RTP activity within the timeout (in seconds), the call will be terminated. This value of RTP Hold Timeout should be larger than RTP Timeout. The default setting is no timeout.
This feature can be used to avoid abnormal call drop when the remote provider requires RTP traffic during proceeding. For example, when the call goes into voicemail and there is no RTP traffic sent out from UCM, configuring this option can avoid voicemail drop. When configured, RTP keep-alive packet will be sent to remote party at the configured interval. If set to 0, RTP keep-alive is disabled.
100rel
Configure the 100rel setting on UCM630X. The default setting is "Yes".
Configure whether the Remote-Party-ID should be trusted. The default setting is Trust Remote Party ID
"No".
Configure whether the Remote-Party-ID should be sent or not. The default Send Remote Party ID
setting is "No".
Generate In-Band Ringing
Configure whether the UCM630X should generate Inband ringing or not. The default setting is "Never". Yes: The UCM630X will send 180 Ringing followed by 183 Session
Progress and in-band audio. No: The UCM630X will send 180 Ringing if 183 Session Progress has not
been sent yet. If audio path is established already with 183 then send inband ringing. Never: Whenever ringing occurs, the UCM630X will send 180 Ringing as long as 200OK has not been set yet. Inband ringing will not be generated even the end point device is not working properly.
Server User Agent
Configure the user agent string for the UCM630X.
Send Compact SIP Headers
Passthrough PAI Header
If enabled, compact SIP headers will be sent. The default setting is "No". Passthrough PAI Header
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SIP Settings/STIR/SHAKEN
To prevent robocalls, UCM now supports STIR/SHAKE protocols. Related options have been added as a new tab in the SIP Settings page.
Clicking on the Add button will show the following window:
Figure 239:SIP Settings/STIR/SHAKEN - Add Authentication Number
Authentication Number Device Name
Creditability
Table 114: SIP Settings/STIR/SHAKEN - Add Authentication Number Settings
Configure the Authentication Number.
Configure the device name. Configure the attestation level, which is the level of confidence of the carrier that the CID has not been spoofed. The following options are available:
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A (Full attestation) - The carrier is associated with the caller and the number. There is high confidence that the CID has not been spoofed.
B (Partial attestation) - The carrier is associated with the caller but not the number. There is uncertainty about whether the CID has been spoofed or not.
C (Gateway attestation) - The carrier is not associated with the caller and has no confidence at all about the number. Generally used for traceback.
Clicking on the Certificate Settings button will bring up the following window:
Figure 240: SIP Settings/STIR/SHAKEN Certificate Settings
Certificate Download Time (s)
Signature Valid Time (s)
Table 115: SIP Settings/STIR/SHAKEN Certificate Settings
Configure the public key download timeout period, the default value is 2 seconds. Configure the validity period of the digital signature, the default value is 15 seconds.
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Private Key Public Key
Configure the Private key. Note: The uploaded file must be less than 2MB in file size, only supports the .key format and must be ECC type. This file will automatically be renamed to "private.key".
Configure the Public Key. Note: The uploaded file must be less than 2MB in file size, only supports the .crt format and must be ECC type. This file will automatically be renamed to "public.crt".
Transparent Call-Info header
UCM supports transparent call info header in order to integrate GDS door system with GXP21XX/GRP261X phones, the UCM will forward the call-info header to the phone in order to request the live view from GDS door system and give the option to open the door via softkey.
Figure 241: Transparent Call-Info
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IAX SETTINGS
The UCM630X IAX global settings can be accessed via Web GUIPBX SettingsIAX Settings.
IAX Settings/General
Bind Port Bind IPv4 Address Bind IPv6 address IAX1 Compatibility
No Checksums
Delay Reject ADSI
Music On Hold Interpret
Music On Hold Suggest Bandwidth
Table 116: IAX Settings/General
Configure the port number that the IAX2 will be allowed to listen to. The default setting is 4569. Force IAX2 to bind to a specific address instead of all addresses. Configure the IPv6 address to bind to. "[::]" means to bind to all IP addresses. Select to configure IAX1 compatibility. The default setting is "No". If selected, UDP checksums will be disabled and no checksums will be calculated/checked on systems supporting this feature. The default setting is "No". If enabled, the IAX2 will delay the rejection of calls to avoid DOS. The default setting is "No". Select to enable ADSI phone compatibility. The default setting is "No". Specify which Music On Hold class this channel would like to listen to when being put on hold. This music class is only effective if this channel has no music class configured and the bridged channel putting the call on hold has no "Music On Hold Suggest" setting. Specify which Music On Hold class to suggest to the bridged channel when putting the call on hold. Configure the bandwidth for IAX settings. The default setting is "Low".
IAX Settings/Registration
Table 117: IAX Settings/Registration
IAX Registration Options
Min Reg Expire
Configure the minimum period (in seconds) of registration. The default setting is 60.
Max Reg Expire
Configure the maximum period (in seconds) of registration. The default setting is 3600.
IAX Thread Count
Configure the number of IAX helper threads. The default setting is 10.
IAX Max Thread Count Configure the maximum number of IAX threads allowed. The default is 100.
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Auto Kill
Authentication Debugging
If enabled and no ACK is received for new messages after the specified wait time, the connection will be terminated.
If enabled, authentication traffic in debugging will not show. The default is "No".
Codec Priority
Configure codec negotiation priority. The default setting is "Reqonly". Caller
Consider the callers preferred order ahead of the host's. Host
Consider the host's preferred order ahead of the caller's. Disabled
Disable the consideration of codec preference all together. Reqonly
This is the same as "Disabled", except when the requested format is not available. The call will only be accepted if the requested format is available.
Type of Service IAX Trunk Options Trunk Frequency Trunk Time Stamps
Configure ToS bit for preferred IP routing.
Configure the frequency of trunk frames (in milliseconds). The default is 20. If enabled, time stamps will be attached to trunk frames. The default is "No".
IAX Settings/Security
Call Token Optional
Table 118: IAX Settings/Static Defense
Enter a single IP address (e.g., 1.1.1.1) or a range of IP addresses (1.1.1.1/255.255.255.255) for which call token validation is not required.
Max Call Numbers
Configure the maximum number of calls allowed for a single IP address.
Max Unvalidated Call Configure the maximum number of Unvalidated calls for all IP addresses.
Numbers
Max Call Numbers
Configure to limit the number of calls for a give IP address of IP range.
IP or IP Range
Enter the IP address (1.1.1.1) or a range of IP addresses (1.1.1.1/255.255.255.255) to be considered for call number limits.
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INTERFACE SETTINGS
Analog Hardware
The analog hardware (FXS port and FXO port) on the UCM630X will be listed in this page. Click on signaling preference for FXS port or configure ACIM settings for FXO port. Select "Loop Start" or "Kewl Start" for each FXS port. And then click on "Update" to save the change.
to edit
Figure 242: FXS Ports Signaling Preference
For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for each port. Or users could click on "Detect" and choose the detection algorithm, two algorithms exist (ERL, Pr) for the UCM630X to automatically detect the ACIM value. The detecting value will be automatically filled into the settings.
Figure 243: FXO Ports ACIM Settings
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Tone Region Advanced Settings FXO Opermode
FXS Opermode
FXS TISS Override
PCMA Override Boost Ringer Fast Ringer Low Power Ring Detect
Table 119: PBX Interface Settings
Select country to set the default tones for dial tone, busy tone, ring tone and etc. to be sent from the FXS port. The default setting is "United States of America (USA)".
Select country to set the On-Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics. The default setting is "United States of America (USA)". Select country to set the On-Hook Speed, Ringer Impedance, Ringer Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics. The default setting is "United States of America (USA)". Configure to enable or disable override Two-Wire Impedance Synthesis (TISS). The default setting is No.
If enabled, users can select the impedance value for Two-Wire Impedance Synthesis (TISS) override. The default setting is 600. Select the codec to be used for analog lines. North American users should choose PCMU. All other countries, unless already known, should be assumed to be PCMA. The default setting is PCMU. Note: This option requires system reboot to take effect.
Configure whether normal ringing voltage (40V) or maximum ringing voltage (89V) for analog phones attached to the FXS port is required. The default setting is "Normal".
Configure to increase the ringing speed to 25HZ. This option can be used with "Low Power" option. The default setting is "Normal".
Configure the peak voltage up to 50V during "Fast Ringer" operation. This option is used with "Fast Ringer". The default setting is "Normal".
If set to "Full Wave", false ring detection will be prevented for lines where Caller ID is sent before the first ring and proceeded by a polarity reversal, as in UK. The default setting is "Standard".
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FXS MWI Mode
FXO Frequency Tolerance
Configure the type of Message Waiting Indicator on FXS lines. The default setting is "FSK". FSK: Frequency Shift Key Indicator NEON: Light Neon Bulb Indicator.
Allows users to adjust the tolerance of the FXO ringing frequency. 63Hz is considered the standard value and is selected by default.
DAHDI Settings
When users encounter issues such as audio delay in outbound calls using the analog trunk, they can adjust DAHDI settings on the UCM to attempt to lessen or resolve the issues.
Figure 244: DAHDI Settings
For the value of the option such as "32, half": The number in the option indicates the number of read/write buffers for TDM (DAHDI). The "Half", "Immediate" or "Full" option indicates the strategy when reading/writing data from buffer.
- "Half": Data will be read/written from buffer when half of the buffer is occupied with data. - "Immediate": Read/write from buffer whenever there is data occupying the buffer. - "Full": Data will be read/written from buffer when buffer is fully occupied with data.
Normally, DAHDI settings should be kept default and should be adjusted only when users encounter analog trunk/Fax-related issues.
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UCM RemoteConnect
An integrated & important part of Grandstream's GDMS cloud-based device management service which runs on Amazon AWS with 99.999% reliability, the UCM RemoteConnect cloud service supports hassle-free Work-FromHome audio/video communications & collaborations using WebRTC-based license-free "Grandstream Wave" soft phones for desktop/Web/mobile devices (plus GUV series of USB headsets/Webcams), zero-touch out-ofbox automated NAT firewall traversal for remote users & devices, IT-friendly remote management of UCM and attached endpoint devices, and more. The RemoteConnect can be configured under Value-added FeaturesRemoteConnect After purchasing the RemoteConnect package.
Figure 245: RemoteConnect
On GDMS platform, sign in and go to DevicePBX Device page, click on "Add Device" to add your UCM6300 device to GDMS system, once done an open beta plan will be assigned to the UCM.
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Note
Figure 246: UCM RemoteConnect - Effective Plan
After the UCM is added on GDMS, automated NAT traversal, SIP extension sync-up and basic statistics features are available without manual configuration required.
Plan Settings
After UCM is added into GDMS, all SIP extensions on the UCM will be synced up to GDMS automatically for users to allocate and manage SIP extension for their end devices. Also, the media NAT Traversal service, alert event sync configuration items are checked by default, the CDR data cloud storage in GDMS should be manually checked according to user needs.
The settings are under UCM webGUIValue-added ServicesUCM RemoteConnectPlan Settings.
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Figure 247: UCM RemoteConnect Plan Settings
Custom logo
Custom logo feature allows users to select a local image file as the new logo. The pictures are in different formats and sizes according to the location of the logo. They are 64*64px (only ico format is supported), 256*256px, 80*80px, which applies for "UCM Login", "Reset Password", "Email Template", "Wave/Login", "Browser Tab interface preview".
LOGO 1: Replaces Browser tab icon LOGO 2: Replaces the Grandstream banner on the top left corner of the management login page and
emails. LOGO 3: Replaces the Grandstream logo on the top left corner of the Wave Web interface and UCM
management interface.
Figure 248: Custom Logo
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Statistics
After using UCM RemoteConnect, all remote calls will be logged and concurrent remote calls will be displayed on the UCM. The concurrent remote calls can be viewed under UCM web GUIValue-Added FeaturesUCM RemoteConnectStatistics page.
Figure 249: Concurrent Remote Calls
For more information, please visit http://ucmrc.gdms.cloud/intro.html and read our UCM63XX RemoteConnect guides
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API CONFIGURATION
The UCM630X supports third party billing interface API for external billing software to access CDR and call recordings on the PBX. The API uses HTTPS to request the CDR data and call recording data matching given parameters as configured on the third-party application.
API Configuration Parameters
Before accessing the API, the administrators need enable API and configure the access/authentication information on the UCM630X first under Value-added FeaturesAPI Configuration. The API configuration parameters are listed in the table below.
Table 120: Configuration Parameters (New)
HTTPS API Settings (New)
Enable
Enable/Disable API. The default setting is enable.
Username
Configure the username for API Authentication.
Password Call Control
Configure the password for API Authentication. If enabled, 3rd party applications will be able to manage inbound calls via API actions. acceptCall will accept incoming calls while refuseCall will reject them. If no actions are done within 10 seconds, calls will automatically be accepted.
API Queries Supported
The new API supports now new queries listed below which will accomplish certain requests and get data about different modules on UCM630X.
Table 121: New API Supported Queries
Queries Supported getSystemStatus
getSystemGeneralStatus listAccount
getSIPAccount updateSIPAccount
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listVoIPTrunk addSIPTrunk getSIPTrunk updateSIPTrunk deleteSIPTrunk listOutboundRoute addOutboundRoute getOutboundRoute updateOutboundRoute deleteOutboundRoute listInboundRoute addInboundRoute getInboundRoute updateInboundRoute deleteInboundRoute playPromptByOrg listBridgedChannels listUnBridgedChannels
Hangup callbarge listQueue getQueue updateQueue addQueue deleteQueue loginLogoffQueueAgent pauseUnpauseQueueAgent listPaginggroup addPaginggroup getPaginggroup updatePaginggroup
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deletePaginggroup MulticastPaging
MulticastPagingHangup listIVR addIVR getIVR
updateIVR deleteIVR
cdrapi recapi pmsapi queueapi getPinSets addPinSets updatePinSets deletePinSets
Table 122: API Configuration Parameters
CDR Real-time Output Settings
Enable
Server Address Port
Enables real-time CDR output module. This module connects to selected IP addresses and ports and posts CDR strings as soon as it is available. CDR server IP address CDR server IP port
Upload Prompts User Configuration
Username
Username used to upload prompts.
Password
Password used to upload prompts.
Upload Voice Prompt via API
Customers now can use the "Upload Prompts User Configuration" to upload/replace voice prompt files as an alternative method to the manual upload method on UCM PBX Settings Voice Prompt Custom Prompt.
The workflow of the prompt file upload goes as:
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An HTTP/HTTPS request is sent to the UCM to upload/replace a voice prompt file, the request should include authentication details to the UCM and the name of the file to be uploaded. Then the UCM will contact an FTP server that should be hosted on the same IP address of the HTTP/HTTPS requester and download the prompt file from the FTP server. The steps and conditions to upload the voice prompt via API are listed below:
1. Configure the prompt User under value-added Features API Configuration Upload Prompts User Configuration. By default, the username and password for voice prompt user are "Username: uploader; Password: uploader123".
Figure 250: Upload Prompt User Configuration
2. Hash the password of the user configured to an MD5 Encryption format. 3. Set the permission on the FTP server to Anonymous on the local computer hosting the FTP server and
make sure that the default FTP port 21 is used. 4. Send an HTTP/HTTPS command to trigger the Prompt file upload on the UCM. If UCM's HTTP server
is set to HTTPS, the example of the request sent to the UCM is:
https://192.168.124.89:8089/cgi?action=uploadprompt&username=uploader&password=9191a6394c2 1b3aabd779213c7179462&filename=test.mp3
If UCM's HTTP server is set to HTTP, the example of the request sent to the UCM is
http://192.168.124.89:8089/cgi?action=uploadprompt&username=uploader&password=9191a6394c21 b3aabd779213c7179462&filename=test.mp3
Note: If the File name on the HTTP/HTTPS request exists already on the UCM's Custom voice prompts list the existing file will be overwritten by the new file downloaded from the FTP server. For more details on CDR API (Access to Call Detail Records) and REC API (Access to Call Recording Files), please refer the document in the link here: http://www.grandstream.com/sites/default/files/Resources/ucm6xxx_cdr_rec_api_guide.pdf
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CTI SERVER
UCM does support CTI server capabilities which are designed to be a part of the CTI solution suite provided by Grandstream, including GXP21XX and GXP17XX enterprise IP phones along with GS Affinity app. Mainly the UCM will by default listening on port TCP 8888 for the connections from GS affinity application in order to interact, modify and serve data requests by the application which includes setting call features for the connected extension as call forward and DND. Users can change the listening port under the menu page, Web GUIValue-added FeaturesCTI Server as shown on below screenshot:
Figure 251: CTI Server Listening port
More information about GS affinity and CTI Support on Grandstream products series please refer to the following link: http://www.grandstream.com/sites/default/files/Resources/GS_Affinity_Guide.pdf
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ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)
The UCM630X supports Asterisk Manager Interface (AMI) with restricted access. AMI allows a client program to connect to an Asterisk instance commands or read events over a TCP/IP stream. It is particularly useful when the system admin tries to track the state of a telephony client inside Asterisk.
User could configure AMI parameters on UCM630X Web GUIValue-added FeaturesAMI. For details on how to use AMI on UCM630X, please refer to the following AMI guide:
http://www.grandstream.com/sites/default/files/Resources/UCM_series_AMI_guide.pdf
---------------------------------------------------------------------------------------------------------------------------------------------------
Warning: Please do not enable AMI on the UCM630X if it is placed on a public or untrusted network unless you have taken steps to protect the device from unauthorized access. It is crucial to understand that AMI access can allow AMI user to originate calls and the data exchanged via AMI is often very sensitive and private for your UCM630X system. Please be cautious when enabling AMI access on the UCM630X and restrict the permission granted to the AMI user. By using AMI on UCM630X you agree you understand and acknowledge the risks associated with this. ---------------------------------------------------------------------------------------------------------------------------------------------------
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CRM INTEGRATION
Customer relationship management (CRM) is a term that refers to practices, strategies and technologies that companies use to manage and analyze customer interactions and data throughout the customer lifecycle, with the goal of improving business relationships with customers. The UCM630X support the following CRMs: SugarCRM, vTigerCRM, ZohoCRM, Salesforce CRM and ACT! CRM, which allows users to look for contact information in the Contacts, Leads and / or Accounts tables, shows the contact record in CRM page, and saves the call information in the contact's history.
SugarCRM
Configuration page of the SugarCRM can be accessed via admin login, on the UCM WebGUIValue-added FeaturesCRM.
Figure 252: SugarCRM Basic Settings
1. Select "SugarCRM" from the CRM System Dropdown in order to use SugarCRM.
CRM System
CRM Server Address Add Unknown Number
Table 123: SugarCRM Settings
Select a CRM system from the dropdown menu, four CRM systems are available: SugarCRM, vTigerCRM, ZohoCRM (v1&v2), Salesforce and ACT! CRM. Enter the IP address of the CRM server. Add the new number to this module if it cannot be found in the selected module.
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Contact Lookups
Select from the "Available" list of lookups and press
to
select where the UCM can perform the lookups on the CRM tables, Leads, Accounts, and Contacts.
Once settings on admin access are configured:
2. Click on
and
.
3. Logout from admin access. 4. Login to the UCM as user and navigate under "User PortalValue-added FeatureCRM User Settings".
Click on "Enable CRM" and enter the username/password associated with the CRM account then click on
and SugarCRM features.
. The status will change from "Logged Out" to "Logged In". User can start then using
Figure 253: CRM User Settings
VTigerCRM
Configuration page of the vTigerCRM can be accessed via admin login, on the UCM WebGUIValue-added FeaturesCRM.
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Figure 254: vTigerCRM Basic Settings
1. Select "vTigerCRM" from the CRM System Dropdown in order to use vTigerCRM.
CRM System
CRM Server Address Add Unknown Number Contact Lookups
Table 124: vTigerCRM Settings
Select a CRM system from the dropdown menu, four CRM systems are available: SugarCRM, vTigerCRM, ZohoCRM (v1&v2), Salesforce and ACT! CRM. Enter the IP address of the CRM server. Add the new number to this module if it cannot be found in the selected module.
Select from the "Available" list of lookups and press
to
select where the UCM can perform the lookups on the CRM tables, Leads, Organizations, and Contacts.
Once settings on admin access are configured:
2. Click on
and
.
3. Logout from admin access. 4. Login to the UCM as user and navigate under "User PortalValue-added FeatureCRM User Settings".
Click on "Enable CRM" and enter the username/password associated with the CRM account then click on
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and SugarCRM features.
. The status will change from "Logged Out" to "Logged In". User can start then using
Figure 255: CRM User Settings
ZohoCRM
Configuration page of the ZohoCRM can be accessed via admin login, on the UCM WebGUIValue-added FeaturesCRM.
Figure 256: ZohoCRM Basic Settings
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1. Select "ZohoCRM" from the CRM System Dropdown in order to use ZohoCRM.
CRM System
CRM Server Address Add Unknown Number Contact Lookups
Table 125: ZohoCRM Settings
Select a CRM system from the dropdown menu, four CRM systems are available: SugarCRM, vTigerCRM, ZohoCRM (v1&v2), Salesforce and ACT! CRM. Enter the IP address of the CRM server. Add the new number to this module if it cannot be found in the selected module.
Select from the "Available" list of lookups and press
to
select where the UCM can perform the lookups on the CRM tables, Leads, Accounts, and Contacts.
Once settings on admin access are configured:
2. Click on
and
.
3. Logout from admin access. 4. Login to the UCM as user and navigate under "User PortalValue-added FeatureCRM User Settings".
Click on "Enable CRM" and enter the username/password associated with the CRM account then click on
and ZohoCRM features.
. The status will change from "Logged Out" to "Logged In". User can start then using
Figure 257: CRM User Settings
Note: ZohoV2CRM is supported as well while the CRM Server Address https://www.zohozpis.com
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Salesforce CRM
Configuration page of the Salesforce CRM can be accessed via admin login, on the UCM Web GUIValueadded FeaturesCRM".
Figure 258: Salesforce Basic Settings
1. Select "Salesforce" from the CRM System Dropdown in order to use Salesforce CRM.
CRM System Add Unknown Number Contact Lookups
Table 126: Salesforce Settings
Select a CRM system from the dropdown menu, four CRM systems are available: SugarCRM, vTigerCRM, ZohoCRM (v1&v2), Salesforce and ACT! CRM. Add the new number to this module if it cannot be found in the selected module.
Select from the "Available" list of lookups and press
to
select where the UCM can perform the lookups on the CRM tables, Leads, Accounts, and Contacts.
Once settings on admin access are configured:
2. Click on
and
.
3. Logout from admin access. 4. Login to the UCM as user and navigate under "User PortalValue-added FeatureCRM User Settings".
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Click on "Enable CRM" and enter the username, password and Security Token associated with the CRM
account then click on
and
. The status will change from "Logged Out" to "Logged In".
User can start then using Salesforce CRM features.
Figure 259: Salesforce User Settings
ACT! CRM
Configuration page of the ACT! CRM can be accessed via admin login, on the UCM Web GUIValue-added FeaturesCRM". The configuration steps of the ACT! CRM are as follows:
1. Navigate to Value-Added FeaturesCRM and select the "ACT! CRM" option.
Figure 260: Enabling ACT! CRM
2. Log into the UCM as a regular user and navigate to Value-Added FeaturesCRM User Settings and check "Enable CRM" option and enter the username and password, which will be the ACT! CRM
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account's API Key and Developer Key, respectively. To obtain these, please refer to the ACT! CRM API developer's guide here: https://mycloud.act.com/act/Help
Figure 261: Enabling CRM on the User Portal
Note: For more information on the ACT! CRM integration, please refer to the ACT! CRM documentation on our website.
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PMS INTEGRATION
UCM630X supports Hotel Property Management System PMS, including check-in/check-out services, wakeup calls, room status, Do Not Disturb which provide an ease of management for hotel applications. This feature can be found on Web GUIValue-added FeaturesPMS. Note: The PMS integration on UCM is currently supported only with one of the three following solutions.
The PMS module built-in the UCM supports the following features based on each solution:
Table 127: PMS Supported Features
Feature
Mitel
HMobile
HSC
IDS
Check-In
Check-out
Wake-up Call
Name Change
Update
Set Credit
Set Station Restriction
Room Status
Room Move
Do Not Disturb
Mini Bar
MSG
MWI
Unconditional Call
Forward
HMobile PMS Connector
In this mode, the system can be divided into three parts: PMS (Property Management System) PMSI (Property Management System Interface) PBX
Grandstream UCM6XXX series have integrated HMobile Connect PMSI which supports a large variety of PMS software providing following hospitality features: Check-in, Check-out, set Room Status, Wake-up call and more. The following figure illustrates the communication flow between the UCM and PMS software, which is done
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through a middleware system (HMobile Connect) acting as interface between both parties. PMS Software
PMSI Middleware (HMobile Connect)
Grandstream UCM
Figure 262: UCM & PMS interaction
HSC PMS
In this mode, the system can be divided into two parts: PMS (Property Management System) PBX
Grandstream UCM6XXX series have integrated HSC PMS providing following features:
Changing Display Name Set Station Restriction Call forwarding DND Name Change MWI Note: 3. Added support for receiving HTTP GET keep-alive messages from HSC PMS. This will allow the PMS
to be aware of its connection to the UCM and take the appropriate actions such as raising alarms, sending notifications, etc. 4. Added support for HTTP GET requests from HSC PMS to retrieve UCM extension information. UCM can provide the following information:
extension UCM extension number
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name extension display name / CID name mwi MWI state permission permission level of the extension cfwt call forwarding always number dnd DND state language display language of the extension in ISO 639-1 format The UCM should respond with either 200 OK or 404 responses. 5. Added HTTPS support The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and PMS software (HSC). The communication between both parties is direct with no middleware.
HSC PMS Software
Grandstream UCM
Figure 263: UCM & HSC PMS interaction
Mitel PMS
In this mode, the system can be divided into two parts: PMS (Property Management System) PBX
Grandstream UCM6XXX series have integrated Mitel PMS providing following hospitality features: Check-in, Check-out, set Room Status, Wake-up call and more. The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and PMS software (Mitel). The communication between both parties is direct with no middleware.
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Mitel PMS Software
Grandstream UCM
Figure 264: UCM & Mitel PMS interaction
IDS PMS
In this mode, the system can be divided into two parts: PMS (Property Management System) PBX
The Grandstream UCM series integrates IDS PMS to set room status, Mini Bar, wake up calls, activate/deactivate dialing permissions, and more.
IDS PMS Software
Grandstream UCM
Figure 265: UCM & IDS PMS interaction
PMS API
The PMS API allows users to use their own middleware to work with PMS systems instead of currently supported integrations. Additionally, this API allows access to read and modify certain UCM parameters that current supported PMS integrations cannot. To use this, users must first enable and configure the HTTPS API settings. For more details, please refer to online HTTPS API, Pmsapi section.
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Connecting to PMS
On the UCM WebGUIValue-added FeaturesPMSBasic Settings" set the connection information for the PMS platform.
Field
PMS Module
Wakeup Prompt PMS URL UCM Port Username Password Site Back Up Voicemail Recordings
Table 128: PMS Basic Settings
Description Users can select the desired PMS module from the drop-down list.
Hmobile. Mitel. HSC. IDS. PMS API. Prompt used when answering the wakeup calls it can be customized from "PBX SettingsVoice PromptCustom Prompt.
Enter the PMS system URL
Enter the Port used by the PMS system Enter the Username to connect to the PMS system
Enter the password to connect to the PMS system Enter PMS site
If enabled, this option allows backing up voicemail recordings to external storage after check-out.
Note: This option is available only when the PMS Module is set to PMS API.
Email address
Email address to send voicemail recordings to upon backup.
In order to use some PMS features please activate the feature code associated under "Call FeaturesFeature Codes"
Update PMS Room Status PMS Wake Up Service
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PMS Features Room Status
User can create Rooms by clicking on "Add Room", the following Figure will be displayed then.
Figure 266: Create New Room
Click "Save" to create the new room, the fields above can be configured from the PMS platform, once set the following screen will be shown:
Figure 267: Room Status
User can create a batch of rooms as well by clicking on
, the following window will pop up:
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Figure 268: Add batch rooms
Wake Up Service In order to create a New Wake up service, user can click on "Add", the following window will pop up:
Field Room Number Select Time Action Status
Figure 269: Create New Wake Up Service
Table 129: PMS Wake up Service
Description Select the room number where to call with a limitation of 63 characters. Set the time of the wakeup call Show the status of the call: Programmed: the call is scheduled for the time set Cancelled: the call is canceled Executed: the wakeup call is made Note: Editing an already executed wakeup service will automatically change the service's status to "Programmed".
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Type
Single: The call will be made once on the specific time. Daily: The call will be repeated every day on the specific time
Once the call is made on the time specified, the following figure show the status of the wakeup call.
Figure 270: Wakeup Call executed
This call has been executed but has been rejected, that why we can see the "Busy" status.
Mini Bar
In order to create a new mini bar, click on "Add Mini Bar" under UCM WebGUIValue-added FeaturesPMSMini Bar, the following window will pop up:
Code
Name Prompt Skip Maid and Password Authentication
Enable Continuous Multi Goods Billing
Figure 271: Create New Mini Bar
Table 130: Create New Mini Bar
Enter a non-existing extension number to be dialed when using the mini bar feature. Enter a name for the mini bar. Select the Prompt to play once connected to the mini bar. If enabled, the default maid code will be 0000, no authentication is required. (Enter 0000 followed by # to access the consumer goods) If enabled, please separate the goods' codes by*.
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In order to create a new maid, click on FeaturesPMSMaid.
under UCM WebGUIValue-added
Maid Code Password
Figure 272: Create New Maid
Table 131: Create New Maid
Enter the Code to use when the maid wants to use the Mini Bar. Enter the password associated with the maid.
In order to create a new consumer goods, click on added FeaturesPMSMini Bar, the following window will popup.
under UCM WebGUIValue-
Code Name
Figure 273: Create New Consumer Goods
Enter the Goods Code. Enter the Name of the Goods
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The Minibar page displays as:
Figure 274: Mini Bar
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WAKEUP SERVICE
The Wake Up service can be used to schedule a reminder or wake up calls to any valid destination. This service is available on the UCM630X as a separated module.
There are three ways to set up Wakeup Service: Using admin login Using user portal Using feature code
Wake Up Service using Admin Login
1. Login to the UCM as admin. 2. Wake Up service can be found under Web GUIValue-added FeaturesWakeup Service, click
on "Add" to create a new wakeup service. The following window will pop up.
Figure 275: Create New Wakeup Service
3. Fill out the required fields and select the members to add to the wakeup group.
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Enable Wakeup Service Name Prompt
Custom Date
Date Time Members
Table 132: Wakeup Service
Enable Wakeup service. Enter a name (up to 64 characters) to identify the wakeup service. Select the prompt to play for that extension.
If disabled, users can select a specific date and time. If enabled users can select multiple days of the week to perform the wakeup.
Select the date or dates when to performs the wakeup call. Select the time when to play the wakeup call. Select the members involved within the wakeup service group.
4. Click
and
to apply the changes.
A wakeup service entry is created. The UCM will send a wakeup call to every extension in the member list at the scheduled date and time.
Note: the wakeup service has the following limitation on how many members can be added depending on UCM model.
UCM Model UCM6301 UCM6302 UCM6304 UCM6308
Table 133: Max Wakeup Members
Max members in a Wakeup Service 50 100 150 200
Wake Up Service from User Portal
1. Login to the user portal on the UCM630X. 2. Wake Up service can be found under "Value-added FeaturesWakeup Service", click on "Add" to
create a new wakeup service. 3. Configures the Name, Prompt, Date and Time for the user to make the wakeup to.
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4. Click
and
to apply the changes.
Wake Up Service using Feature Code
1. Login to the UCM as admin. 2. Enable "Wakeup Service" from the WebGUI under "Call FeaturesFeature Codes".
3. Click
and
to apply the changes.
4. Dial "*36" which is the feature code by default to access to the UCM wakeup service to add, update, activate or deactivate UCM wakeup service.
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ANNOUNCEMENTS CENTER
The UCM630X supports Announcements Center feature which allows users to pre-record and store voice message into UCM630X with a specified code. The users can also create group with specified extensions. When the code and the group number are dialed together in the combination of code + group number, the specified voice message is sent to all group members and only extensions in the group will hear the voice message.
Figure 276: Announcements Center
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Announcements Center Settings
Name
Code
Custom Prompt Ring Timeout Auto Answer
Table 134: Announcements Center Settings
Configure a name for the newly created Announcements Center to identify this announcement center.
Enter a code number for the custom prompt. This code will be used in combination with the group number. For example, if the code is 55, and group number is 666. The user can dial 55666 to send prompt 55 to all members in group 666. Note: The combination number must not conflict with any number in the system such as extension number or conference number. This option is to set a custom prompt as an announcement to notify group members. The file can be uploaded from page `Custom Prompt'. Click `Prompt' to add additional record. Configure the ring timeout for the group members. The default value is 30 seconds. If set to Yes the Auto answer will be enabled by the members.
Group Settings
Name Number Members
Table 135: Group Settings
Configure a name for the newly created group to identify the group. Note: Name cannot exceed 64 characters. Configure the group number. The group number is used in combination with the code. For example, if group number is 666, and code is 55. The user can dial 55666 to send prompt 55 to all members in group 666. Note: The combination number must not conflict with any number in the system such as extension number or conference number and cannot exceed 64 characters. Select the group members from the available list.
Announcements Center feature can be found under Web GUIValue-added FeaturesAnnouncements Center. The following example demonstrates the usage of this feature.
1. Click
to add new group.
2. Give a name to the newly created group.
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3. Create a group number which is used with code to send voice message. 4. Select the extensions to be included in the group, who will receive the voice message.
Figure 277: Announcements Center Group Configuration
In this example, group "Test" has number 666. Extension 1000, 1001 and 1002 are in this group.
1. Click
to create a new Announcement Center.
2. Give a name to the newly created Announcement Center. 3. Specify the code which will be used with group number to send the voice message to. 4. Select the message that will be used by the code from the Custom Prompt drop down menu. To create a
new Prompt, please click "Prompt" link and follow the instructions in that page.
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Figure 278: Announcements Center Code Configuration
Code and Group number are used together to direct specified message to the target group. All extensions in the group will receive the message. For example, we can send code 55 to group 666 by dialing 55666 from any extension registered to the UCM630X. All the members in group 666 which are extension 1000, 1001 and 1002 will receive this voice message after they pick up the call.
Figure 279: Announcements Center Example
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STATUS AND REPORTING
PBX Status
The UCM630X monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces and Parking lot. It presents administrators the real-time status in different sections under Web GUISystem StatusDashboard.
Figure 280: StatusPBX Status
Trunks Users could see all the configured trunk status in this section.
Figure 281: Trunk Status
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Table 136: Trunk Status
Display trunk status.
Status
Analog trunk status: Available Busy Unavailable Unknown Error
SIP Peer trunk status: Unreachable: The hostname cannot be reached. Unmonitored: Heartbeat feature is not turned on to be monitored. Reachable: The hostname can be reached.
Trunks
Type
Username Port/Hostname/IP
SIP Register trunk status: Registered Unrecognized Trunk
Display trunk name Display trunk Type: Analog SIP IAX Display username for this trunk. Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.
Extensions
Extensions Status can be seen from the same configuration page, users can go under Web GUIExtension/TrunkExtensions and following page will be displayed listing the extensions and their status information.
Figure 282: Extension Status
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Status
Presence Status Extension Name IP and Port Email
Table 137: Extension Status
Display extension number (including feature code). The color indicator has the following definitions.
Green: Free
Blue: Ringing
Yellow: In Use
Grey: Unavailable
Display the presence status of the extension.
Display the extension number.
First name and last name of the extension.
Display the IP and port number of the registered device.
Display Email Notification status for the extension.
When notification is waiting for be sent, shows and once sent it will display
Terminal Type
Displays extension type. SIP User IAX User Analog User Ring Groups Voicemail Groups
Interfaces Status
This section displays interface/port connection status on the UCM630X. The following example shows the interface status for UCM6304 with USB, WAN port, FXS1, FXS2 and FXO1 connected.
Figure 283: UCM6304 Interfaces Status
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Table 138: Interface Status Indicators
USB connected. USB disconnected. SD Card connected. SD Card disconnected. LAN/WAN connected. LAN/WAN not configured. LAN/WAN disconnected. FXS/FXO connected. FXS/FXO waiting. FXS/FXO busy. FXS/FXO not configured. FXS/FXO disconnected.
System Status
The UCM630X system status can be accessed via Web GUIStatusSystem Status, which displays the following system information.
General Under Web GUISystem StatusSystem InformationGeneral, users could check the hardware and software information for the UCM630X. Please see details in the following table.
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Table 139: System StatusGeneral
System Status System InformationGeneral
Model
Product model.
Part Number
Product part number.
System Time
Current system time. The current system time is also available on the upper right of each web page.
Up Time
System up time since the last reboot.
Boot
Boot version.
Core
Core version.
Base
Base version.
Program
Program version. This is the main software release version.
Recovery
Recovery version.
Lang
Lang version
GSWave
GSWave version
Network
Under Web GUISystem StatusSystem InformationNetwork, users could check the network information for the UCM630X. Please see details in the following table.
Table 140: System StatusNetwork
System StatusSystem StatusNetwork
MAC Address
Global unique ID of device, in HEX format. The MAC address can be found on the label coming with original box and on the label located on the bottom of the device.
IPv4 Address
IPv4 address.
IPv6 Address Link IPv6 address
Gateway
Default gateway address.
Subnet Mask
Subnet mask address.
DNS Server
DNS Server address.
Duplex Mode
Duplex Mode
Speed
Speed
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Storage Usage Users could access the storage usage information from Web GUISystem StatusDashboardStorage Usage. It shows the available and used space for Space Usage and Inode Usage.
Space Usage includes:
Configuration partition This partition contains PBX system configuration files and service configuration files.
Data partition Voicemail, recording files, IVR file, Music on Hold files etc.
USB disk USB disk will display if connected.
SD Card SD Card will display if connected.
Inode Usage includes:
Configuration partition Data partition
Note: Inode is the pointer used for file reference in the system. The system usually has limited resources of pointers
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Figure 284: System StatusStorage Usage
Resource Usage When configuring and managing the UCM630X, users could access resource usage information to estimate the current usage and allocate the resources accordingly. Under Web GUI System Status Dashboard Resource Usage, the current CPU usage and Memory usage are shown in the pie chart.
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Figure 285: System StatusResource Usage
System Events
The UCM630X can monitor important system events, log the alerts, and send Email notifications to the system administrator.
Alert Events List
The system alert events list can be found under Web GUI Maintenance System Events. The following event and their actions are currently supported on the UCM630X which will have alert and/or Email generated if occurred:
Table 141: Alert Events
Action index
Alert Events
1
Disk Usage
2
Modify Super Admin Password
3
Memory Usage
4
System Reboot
5
System Update
6
System Crash
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7
Register SIP failed
8
Register SIP trunk failed
9
Restore Config
10
User login success
11
User login failed
12
SIP Internal Call Failure
13
SIP Outgoing Call through Trunk Failure
14
Fail2ban Blocking
15
SIP Lost Registration
16
SIP Peer Trunk Status
17
User Login Banned
18
HA failure warning
19
Emergency Calls
20
The CDR database is corrupted
21
NAS
22
Data Sync Backup
23
Remote Concurrent Calls
24
External Disk Status
25
CPU Usage Call Control
Figure 286: Alert Event List
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Note: For users who have purchased a GDMS package, once the option Alert Events Sync is enabled under RemoteConnect, the triggered events will be pushed to their GDMS platform. For more information, please refer to: http://www.grandstream.com/sites/default/files/Resources/UCM63xx_RemoteConnect_User_Guide.pdf
Click on
to configure the parameters for each event. See examples below.
1. Disk Usage
Figure 287: System EventsAlert Events Lists: Disk Usage
Detect Cycle: The UCM630X will perform the internal disk usage detection based on this cycle. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM630X system will send the alert.
Note: If the threshold is exceeded, any behavior of operating the disk will be rejected, including stopping file upload, IM writing, recording and CDR recording.
2. External Disk Usage
Figure 288: System EventsAlert Events Lists: External Disk Usage
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Detect Cycle: The UCM630X will perform the External disk usage detection based on this cycle. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM630X system will send the alert.
3. Memory Usage
Figure 289: System EventsAlert Events Lists: Memory Usage
Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM630X system will send the alert.
4. System Crash
Figure 290: System EventsAlert Events Lists: System Crash
Detect Cycle: The UCM will detect the event at each cycle based on the specified time. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
Click on the switch
to turn on/off the alert and Email notification for the event. Users could
also select the checkbox for each event and then click on button "Alert On", "Alert Off", "Email Notification On", "Email Notification Off" to control the alert and Email notification configuration.
5. Modify Super Admin Password
Once the super administrator password is modified, the system will record the password modification event in the alarm log.
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6. System Reboot UCM will detect the system restart and will send an alert for it. 7. System Reboot Once the system is upgraded, the system upgrade event will be recorded in the alarm log. 8. SIP registration failed
Figure 291:System EventsAlert Events Lists: Register SIP Failed
Configure the sending period of the SIP registration failure alert. The first registration failure alert of the same IP to the same SIP account will be sent immediately, and then no alerts will be sent for similar failure warnings in the cycle time. After the cycle time expires, an alert will be sent again to count the number of occurrences of similar SIP registration failure alerts during the cycle. When set to 0, alerts are always sent immediately.
9. Register SIP trunk failed
Figure 292: System EventsAlert Events Lists: Register SIP Trunk Failed
Detect Cycle: The UCM will detect the failure of SIP trunk registration at a set interval. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
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10. Restore Config Once the system configuration is restored, the configuration restoration event will be recorded in the alert log. 11. User login success Successful user login events will be recorded in the alert log. 12. User login failed User login failure events will be recorded in the alert log. 13. SIP Internal Call Failure If the system SIP extension call fails within the office, the event will be recorded in the alert log. 14. SIP Outgoing Call through Trunk Failure If the system SIP trunk outgoing call fails, the event will be recorded in the alert log. 15. Fail2ban blocking If the system Fail2ban is blocking, the event will be recorded in the alert log. 16. SIP lost registration If System SIP extension registration is lost, the event will be recorded in the alert log. 17. SIP peer trunk status If the SIP peer trunks status is abnormal, the event will be recorded in the alert log. 18. User login banned If user login is blocked, the event will be recorded in the alert log. 19. Emergency Calls If the system generates an emergency call, the event will be recorded in the alert log. 20. NAS
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If the system network disk is abnormal, the event will be recorded in the alarm log. 21. Data Sync Backup If the system performs data synchronization and backup abnormalities, the event will be recorded in the alert log. 22. Remote concurrent calls If the remote concurrent call fails, the event will be recorded in the alert log. 23. External disk status If the external disk of the system is Connected/Disconnected, the event will be recorded in the alarm log. 24. CPU Flow control The CPU flow control threshold is defined under System Settings -> General Settings, and the default value is 90%. When the traffic exceeds the predetermined value, the event will be recorded in the alert log and new calls will be prohibited.
Alert Log Under Web GUIMaintenanceSystem EventsAlert Log, system messages from triggered system events are listed as alert logs. The following screenshot shows system crash alert logs.
Figure 293: System EventsAlert Log
User could also filter alert logs by selecting a certain event category, type of alert log, and/or specifying a certain
time period. The matching results will be displayed after clicking on types by the system:
. Alert logs are classified into two
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1. Generate Alert: Generated when alert events happen, for example, alert logs for disk usage exceeding the alert threshold.
2. Restore to Normal: Generated when alert events being cleared, for example, logs for disk usage dropping back below the alert threshold.
User could filter out alert logs of "Generate Alert" or "Restore to Normal" by specifying the type according to need. The following figure shows an example of filtering out alert logs of type of "Restore to Normal".
Figure 294: Filter for Alert Log
Alert Contact
This feature allows the administrator to be notified when one of the Alert events mentioned above happens. Users could add administrator's Email address under Web GUIMaintenanceSystem EventsAlert Contact to send the alert notification to an email (Up to 10 Email addresses can be added) or also specify an HTTP server where to send this alert.
Super Admin Email Admin Email
Email Template
Table 142: Alert Contact
Configure the email addresses to send alert notifications to. Up to 10 email addresses can be added. Configure the email addresses to send alert notifications to. Up to 10 email addresses can be added. Please refer to section Email Templates
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Protocol
Protocol used to communicate with the server. HTTP or HTTPS. Default one is HTTP.
HTTP Server
The IP address or FQDN of the HTTP/HTTPS server.
HTTP Server Port HTTP/HTTPS port
Warning Template Customize the template used for system warnings. By default: {"action":"${ACTION}","mac":"${MAC}","content":"${WARNING_MSG}"}
Notification Template
Customize the notification template to receive relevant alert information. By default: {"action":"${ACTION}","cpu":"${CPU_USED}","memery":"${MEM_USED}","disk":"${ DISK_USED}","external_disk":"${EXTERNAL_DISK_USED}"} Note: The notification message with "action:0" will be sent periodically if Notification Interval is set.
Notification Interval
Modifies the frequency at which notifications are sent in seconds. No notifications will be sent if the value is "0". Default value: 20
Template Variables
${MAC} : MAC Address ${WARNING_MSG} : Warning message ${TIME} : Current System Time ${CPU_USED} : CPU Usage ${MEM_USED} : Memory Usage ${ACTION} : Message Type. Refer to [Table 141: Alert Events] ${DISK_USED} : Disk Usage ${EXTERNAL_DISK_USED} : Disk Usage
CDR
CDR (Call Detail Record) is a data record generated by the PBX that contains attributes specific to a single instance of phone call handled by the PBX. It has several data fields to provide detailed description for the call, such as phone number of the calling party, phone number of the receiving party, start time, call duration, etc.
On the UCM630X, the CDR can be accessed under Web GUICDRCDR. Users could filter the call report by specifying the date range and criteria, depending on how the users would like to include the logs to the report. Click on "Filter" button to display the generated report.
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Figure 295: CDR Filter
Call Type
Table 143: CDR Filter Criteria
Groups the following:
Inbound calls: Inbound calls are calls originated from a non-internal source (like a VoIP trunk) and sent to an internal extension.
Outbound calls: Outbound calls are calls sent to a non-internal source (like a VoIP trunk) from an internal extension.
Internal calls: Internal calls are calls from one internal extension to another extension, which are not sent over a trunk.
External calls: External calls are calls sent from one trunk to another trunk, which are not sent to any internal extension.
Status
Filter with the call status, the available statuses are the following:
Answered No Answer Busy Failed
Source Trunk Name
Select source trunk(s) and the CDR of calls going through inbound the trunk(s) will be filtered out.
Destination Trunk Select destination trunk(s) and the CDR of calls going outbound through the trunk(s)
Name
will be filtered out.
Action Type
Filter calls using the Action Type, the following actions are available:
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Extension Group Export File Data
Announce Announcement page Dial Announcements Callback Call Forward Conference DISA Follow Me IVR Page Parked Call Queue Ring Group Transfer VM VMG Video Conference VQ_Callback Wakeup Emergency Call Emergency Notify SCA Specify the Extension Group name to filter with.
Select the fields that will be exported, the following fields are available:
Account Code Session Premier caller Action type Source trunk name Destination trunk name Caller number Caller ID Caller name Callee number Answer by Context Start time
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Account Code Start Time End Time Caller Number
Caller Name Callee Number
Answer time End time Call time Talk time Source channel Dest channel Call status Dest channel extension Last app Last data AMAFLAGS UIQUEID Call type NAT Select the account Code to filter with. If pin group CDR is enabled, the call with pin group information will be displayed as part of the CDR under Account Code Field. Specify the start time to filter the CDR report. Click on the calendar icon on the right and the calendar will show for users to select the exact date and time. Specify the end time to filter the CDR report. Click on the calendar icon on the right and the calendar will show for users to select the exact date and time. Enter the caller number to filter the CDR report. CDR with the matching caller number will be filtered out. User could specify a particular caller number or enter a pattern. `.' matches zero or more characters, only appears in the end. `X' matches any digit from 0 to 9, caseinsensitive, repeatable, only appears in the end.
For example: 3XXX: It will filter out CDR that having caller number with leading digit 3 and of 4 digits' length. 3.: It will filter out CDR that having caller number with leading digit 3 and of any length. Enter the caller name to filter the CDR report. CDR with the matching caller name will be filtered out. Enter the callee number to filter the CDR report. CDR with the matching callee number will be filtered out. Note: The "Callee Number" filter field supports specifying Pattern (example: 3XXX) or using Leading digits (example: 3.) as filtering options.
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The call report will display as the following figure shows.
Figure 296: Call Report
The CDR report has the following data fields:
Start Time Format: 2019-12-11 09:53:03
Action Type Example: IVR DIAL WAKEUP
Call From Example format: 5555
Call To Example format: 1000
Call Time Format: 0:00:11
Talk Time Format: 0:00:06
Account Code Example format: Grandstream/Test
Status Answered, Busy, No answer or Failed.
Users could perform the following operations on the call report.
Sort by "Start Time" Click on the header of the column to sort the report by "Start Time". Clicking on "Start Time" again will
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reverse the order.
Download Searched Results Click on "Download Search Result(s)" to export the records filtered out to a .csv file.
Download All Records Click on "Download All Records" to export all the records to a .csv file.
Delete All Click on
button to remove all the call report information.
Delete Search Result
On the bottom of the page, click on
button to remove CDR records that
appear on search results.
Note: When deleting CDR, a prompt will now appear asking whether to delete all recording files or not.
Play/Download/Delete Recording File (per entry) If the entry has audio recording file for the call, the three icons on the rightest column will be activated for users to select. In the following picture, the second entry has audio recording file for the call.
Click on
to play the recording file; click on to download the recording file in .wav format; click on
to delete the recording file (the call record entry will not be deleted).
Figure 297: Call Report Entry with Audio Recording File
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Automatic Download CDR Records User could configure the UCM630X to automatically download the CDR records and send the records to multiple Email recipients in a specific hour. Click on "Automatic Download Settings" and configure the parameters in the dialog below.
Figure 298: Automatic Download Settings
To receive CDR record automatically from Email, check "Enable" and select a time period "By Day" "By Week" or "By Month", select Hour of the day as well for the automatic download period. Make sure you have entered an Email or multiple email addresses where to receive the CDR records.
Note: users have the option to delete the sent records "Delete Sent Records" Starting from UCM630X firmware 1.0.10.x, transferred call will no longer be displayed as a separate call entry in CDR. It will display within call record in the same entry. CDR new features can be found under Web GUICDRCDR. The user can click on the option icon for a specific call log entry to view details about this entry, such as premier caller and transferred call information.
Figure 299: CDR Report
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Figure 300: Detailed CDR Information
Downloaded CDR File The downloaded CDR (.csv file) has different format from the Web GUI CDR. Here are some descriptions. Caller number, Callee number "Caller number": the caller ID. "Callee number": the callee ID.
If the "Source Channel" contains "DAHDI", this means the call is from FXO/PSTN line.
Figure 301: Downloaded CDR File Sample
Context There are different context values that might show up in the downloaded CDR file. The actual value can vary case by case. Here are some sample values and their descriptions. from-internal: internal extension makes outbound calls. ext-did-XXXXX: inbound calls. It starts with "ext-did", and "XXXXX" content varies case by case, which also relate to the order when the trunk is created. ext-local: internal calls between local extensions.
Source Channel, Dest Channel Sample 1:
Figure 302: Downloaded CDR File Sample - Source Channel and Dest Channel 1
DAHDI means it is an analog call, FXO or FXS. For UCM6302, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports. For UCM6304, DAHDI/(1-4) are FXO ports, and DAHDI(5-6) are FXS ports. For UCM6308, DAHDI/(1-8) are FXO ports, and DAHDI(9-10) are FXS ports.
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Sample 2:
Figure 303: Downloaded CDR File Sample - Source Channel and Dest Channel 2
"SIP" means it is a SIP call. There are three format: (a) PJSIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string and can be ignored. (c) PJSIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out through the trunk. (c) PJSIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this trunk. The last XXXXX is a random string and can be ignored. There are some other values, but these values are the application name which are used by the dialplan.
IAX2/NUM-XXXXXXX: it means this is an IAX call. Local/@from-internal-XXXXX: it is used internally to do some special feature procedure. We can simply ignore it. Hangup: the call is hung up from the dialplan. This indicates there are some errors or it has run into abnormal cases. Playback: play some prompts to you, such as 183 response or run into an IVR. ReadExten: collect numbers from user. It may occur when you input PIN codes or run into DISA
Note: The language of column titles in exported CDR reports and statistics reports will be based on the UCM's display language
CDR Export Customization
Users can select the data they want to see in exported CDR reports by first clicking on the Filter button on the CDR page under CDRCDR and selecting the desired information in the Export File Data field.
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Figure 304: CDR Export File data
CDR in GDMS Cloud Cloud Storage for CDR Record which can be displayed under CDR CDR in GDMS Cloud.
Figure 305: CDR in GDMS Cloud
Statistics CDR Statistics is an additional feature on the UCM630X which provides users a visual overview of the call report across the time frame. Users can filter with different criteria to generate the statistics chart.
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Trunk Type Call Type Time Range
Figure 306: CDR Statistics
Table 144: CDR Statistics Filter Criteria
Select one of the following trunk type. All SIP Calls PSTN Calls Select one or more in the following checkboxes. Inbound calls Outbound calls Internal calls External calls All calls By month (of the selected year). By week (of the selected year). By day (of the specified month for the year). By hour (of the specified date). By range. For example, 2016-01 To 2016-03.
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Recording Files
This page lists all the recording files recorded by "Auto Record" per extension/ring group/call queue/trunk, or via feature code "Audio Mix Record". If external storage device is plugged in, for example, SD card or USB drive, the files are stored on the external storage. Otherwise, internal storage will be used on the UCM630X.
Figure 307: CDRRecording Files
Click on "Delete Selected Recording Files" to delete the recording files. Click on "Delete All Recording Files" to delete all recording files. Click on "Batch Download Recording Files" in order to download the selected recording files. Click on "Download All Recording Files" to download all recordings files. Select Either "USB Disk" or "Local" to show recording files stored on external or internal storage, depending
on selected storage space. Select whether to show call recordings, queue recordings or conference recordings.
Click on to download the recording file in .wav format.
Click on
to delete the recording file.
To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding column. Click on the title again can switch the sorting mode between ascending order or descending order.
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USER PORTAL
Users could log into their web GUI portal using the extension number and user password. When an extension is created in the UCM630X, the corresponding user account for the extension is automatically created. The user portal allows access to a variety of features which include user information, extension configuration and CDR as well as settings and managing value-added features like Call Queue, Wakeup Service and CRM.
Users also can access their personal data files (call recordings, Voicemail Prompts ...).
The login credentials are configured by Super Admin. The following figure shows the dialog of editing the account information by Super Admin. The Username must be the extension number and it is not configurable, and the password is set on "User Password" field and it should not be confused with the SIP extension password.
Figure 308: Edit User Information by Super Admin
The following screenshot shows an example of login page using extension number 1000 as the username.
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Figure 309: User Portal Login
After login, the Web GUI display is shown as below.
Figure 310: User Portal Layout
After successful login, the user has the following three configuration tabs:
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Basic Information
Under this menu, the user can configure and change his/her personal information including (first name, last name, password, email address, department...). And they can also set and activate their extension features (presence status, call forward, DND ....) to be reflected on the UCM. Also, the user can see from this menu the Call Details Records and search for specific ones along with the possibility to download the records on CSV format for later usage.
Personal Data
Under this section, the user can access and manage their personal data files which includes (voicemail files, call recordings ...) along with the possibility to set Follow me feature to without requesting the Super admin to set the feature from admin account.
Value-added Features
On this section, the user has access to manage and use all rich value-added features which includes.
+ If user is a member of call queue, they can check the queue's activity from the "Call Queue" section. + Create and enable Wake Up service. + Enable and configure CRM connection to either SugarCRM or Salesforce.
For the configuration parameter information in each page, please refer to [Table 145: User ManagementCreate New User] for options in User PortalBasic InformationUser Information page; please refer to [EXTENSIONS] for options in User PortalBasic InformationExtension page; please refer to [CDR] for User PortalBasic InformationCDR page.
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MAINTENANCE
User Management
User management is on Web GUIMaintenanceUser Management page. User could create multiple accounts for different administrators to log in the UCM630X Web GUI. Additionally, the system will automatically create user accounts along with creating new extensions for extension users to login to the Web GUI using their extension number and password. All existing user accounts for Web GUI login will be displayed on User Management page as shown in the following figure.
Figure 311: User Management Page Display
User Information When logged in as Super Admin, click on "Add" to create a new account for Web GUI user. The following dialog will prompt. Configure the parameters as shown in below table.
Figure 312: Create New User
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Username
User Password
Privilege Department Email Address First Name Last Name Home Number Phone Number
Table 145: User ManagementCreate New User
Configure a username to identify the user which will be required in Web GUI login. Letters, digits, and underscore are allowed in the username. Configure a password for this user which will be required in Web GUI login. English input is allowed without space,' and ". This is the role of the Web GUI user. Currently only "Admin" is supported when Super Admin creates a new user.
Enter the necessary information to keep a record for this user.
Once created, the Super Admin can edit the users by clicking on
or delete the user by clicking on .
Figure 313: User Management New Users
Custom Privilege
Four privilege levels are supported:
Super Administrator
- This is the highest privilege. Super Admin can access all pages on UCM630X Web GUI, change configuration for all options and execute all the operations.
- Super Admin can create, edit, and delete one or more users with "Admin" privilege - Super Admin can edit and delete one or more users with "Consumer" privilege - Super Admin can view operation logs generated by all users. - By default, the user account "admin" is configured with "Super Admin" privilege and it is the only user
with "Super Admin" privilege. The Username and Privilege level cannot be changed or deleted. - Super Admin could change its own login password on Web GUIMaintenanceLogin Settings page.
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- Super Admin could view operations done by all the users in Web GUIMaintenanceUser ManagementOperation Log
Administrator - Users with "Admin" privilege can only be created by "Super Admin" user. - "Admin" privilege users are not allowed to access the following pages: MaintenanceUpgrade MaintenanceCleaner MaintenanceReset/Reboot SettingsUser ManagementOperation Log - "Admin" privilege users cannot create new users for login.
Note: By default, administrator accounts are not allowed to access backup menu, but this can be assigned
to them by editing the option "Maintenance User Management Custom Privilege" then press to edit the "Admin" account and include backup operation permission for these types of users.
Consumer
Figure 314: Assign Backup permission to "Admin" users
- A user account for Web GUI login is created automatically by the system when a new extension is created.
- The user could log in the Web GUI with the extension number and password to access user information, extension configuration, CDR of that extension, personal data, and value-added features. For more details; please refer to User Portal Guide.
- The SuperAdmin user can click on on the "General_User" in order to enable/disable the custom privilege from deleting their own recording files, changing SIP credentials, and disabling voicemail service in their user portal account.
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Figure 315: General User
Custom Privilege
The Super Admin user can create users with different privileges. 33 items are available for privilege customization.
- API Configuration - API Configuration - Backup - Callback - Call Queue - CDR Recording Files - CDR Records - CDR Statistics - Audio Conference - Dial By Name - DISA - Emergency Calls - Event List - Extensions - Outbound Routes - Inbound Routes - Feature Codes - IVR - Paging/Intercom - Parking Lot - Pickup Groups
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- PMS - Wakeup Service - Ring Groups - SCA - Speed Dial - System Status - System Events - Time Settings - Video Conference - Voicemail - Voice Prompt - Wakeup Service - Zero Config - LDAP Server - UCM RemoteConnect - Announcement.
Figure 316: Create New Custom Privilege
Log in UCM630X as super admin and go to MaintenanceUser ManagementCustom Privilege, create privilege with customized available modules.
To assign custom privilege to a sub-admin, navigate to UCM Web GUIMaintenanceUser ManagementUser InformationCreate New User/Edit Users, select the custom privilege from "Privilege" option.
Concurrent Multi-User Login
When there are multiple Web GUI users created, concurrent multi-user login is supported on the UCM630X. Multiple users could edit options and have configurations take effect simultaneously. However, if different users are editing the same option or making the same operation (by clicking on "Apply Changes"), a prompt will pop up as shown in the following figure.
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Figure 317: Multiple User Operation Error Prompt
Change Password
After logging in the UCM630X Web GUI for the first time, it is highly recommended for users to change the default password to a more complicated password for security purpose. Follow the steps below to change the Web GUI access password.
1. Go to Web GUIMaintenanceLogin SettingsChange Password / Email page. 2. Enter the old password first. 3. Enter the new password and re-type the new password to confirm. The new password has to be at least 4
characters. The maximum length of the password is 30 characters. 4. Configure the Email Address that is used when login credential is lost. 5. Click on "Save" and the user will be automatically logged out. 6. Once the web page comes back to the login page again, enter the username "admin" and the new password
to login.
Figure 318: Change Password
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Enter Old Password Change Password Enter New Password Re-enter New Password Change Username Please enter the username
Email Address
Enter the Old Password for UCM630X Enable Change Password Enter the New Password for UCM630X Retype the New Password for UCM630X Enable Change Username Enter the Username The Email address is the User Email Address. It is used for receiving password information if the user forgets his password.
Change Username
UCM630X allows users now to change Super Administrator username.
Figure 319: Change Username
Change binding Email
UCM630X allows user to configure binding email in case login password is lost. UCM630X login credential will be sent to the designated email address. The feature can be found under Web GUI MaintenanceLogin SettingsChange Password / Email
Email Address
Figure 320: Change Binding Email Table 146: Change Binding Email option
Email Address is used to retrieve password when password is lost
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Login Security
After the user logs in the UCM630X Web GUI, the user will be automatically logged out after certain timeout, or he/she can be banned for a specific period if the login timeout is exceeded. Those values can be specified under UCM630X web GUIMaintenanceLogin SettingsLogin Security page.
The "User Login Timeout" value is in minute and the default setting is 10 minutes. If the user does not make any operation on Web GUI within the timeout, the user will be logged out automatically. After that, the Web GUI will be redirected to the login page and the user will need to enter username and password to log in. If set to 0, there is no timeout for the Web GUI login session and the user will not be automatically logged out.
"Maximum number of login attempts" can prevent the UCM630X from brutal force decryption, if this number is exceeded user IP address will be banned from accessing the UCM for a period of time based on user configuration, the default value is 5.
"User ban period" specify the period of time in minutes an IP will be banned from accessing the UCM if the User max number of try login is exceeded, the default value is 5.
"Login Banned User List" show the list of IPs' banned from the UCM.
"Login Whitelist" User can add a list of IPs' to avoid the above restriction, thus, they can exceed the User max number of try login.
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Operation Log
Figure 321: Login Timeout Settings
Super Admin has the authority to view operation logs on UCM630X Web GUISettingsUser ManagementOperation Log page. Operation logs list operations done by all the Web GUI users, for example, Web GUI login, creating trunk, creating outbound rule and etc. There are 7 columns to record the operation details "Date", "Username", "IP Address", "Results", "Page Operation", "Specific Operation" and "Remark".
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Figure 322: Operation Logs
The operation log can be sorted and filtered for easy access. Click on or at the top of each column to sort. For example, clicking on for "Date" will sort the logs according to newer operation date and time. Clicking on for "Date" will reverse the order.
Date Username IP Address Results
Page Operation
Table 147: Operation Log Column Header
The date and time when the operation is executed. The username of the user who performed the operation. The IP address from which the operation is made. The result of the operation.
The page where the operation is made. For example, login, logout, delete user, create trunk and etc.
Specific Operation
Click on to view the options and values configured by this operation.
Remark
Allows users to add notes and remarks to each operation
User could also filter the operation logs by time condition, IP address and/or username. Configure these conditions and then click on "Display Filter".
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Figure 323: Operation Logs Filter
The above figure shows an example that operations made by user "support" on device with IP 192.168.40.173 from 2014-11-01 00:00 to 2014-11-06 15:38 are filtered out and displayed.
To delete operation logs, users can perform filtering first and then click on
to delete
the filtered result of operation logs. Or users can click on
to delete all operation logs at once.
Upgrading
The UCM630X can be upgraded to a new firmware version locally. And in order to do that, please follow the below steps: 1. Download the latest UCM630X firmware file from the following link and save it in your PC.
http://www.grandstream.com/support/firmware 2. Log in the Web GUI as administrator in the PC. 3. Go to Web GUIMaintenanceUpgrade, upload the firmware file by clicking on "choose file to upload"
and select the firmware file from your PC. The default firmware file name is UCM630Xfw.bin
Figure 324: Local Upgrade
Figure 325: Upgrading Firmware Files
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4. Wait until the upgrading process is successful and a window will be popped up in the Web GUI.
Figure 326: Reboot UCM630X
5. Click on "OK" to reboot the UCM630X and check the firmware version after it boots up.
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Notes: - Please do not interrupt or power cycle the UCM630X during upgrading process. - The firmware file name allows the use of the special characters besides the following restricted characters: # $ ^ & * + ( ) [ ] / ; ` | , < > ?
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No Local Firmware Servers
Service providers should maintain their own firmware upgrade servers. For users who do not have TFTP/HTTP/HTTPS server, some free windows version TFTP servers are available for download from http://www.solarwinds.com/products/freetools/free_tftp_server.aspx http://tftpd32.jounin.net
Please check our website at http://www.grandstream.com/support/firmware for latest firmware.
Instructions for local firmware upgrade via TFTP:
1. Unzip the firmware files and put all of them in the root directory of the TFTP server; 2. Connect the PC running the TFTP server and the UCM630X to the same LAN segment; 3. Launch the TFTP server and go to the File menuConfigureSecurity to change the TFTP server's default
setting from "Receive Only" to "Transmit Only" for the firmware upgrade; 4. Start the TFTP server and configure the TFTP server in the UCM630X web configuration interface; 5. Configure the Firmware Server Path to the IP address of the PC; 6. Update the changes and reboot the UCM630X. End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use Microsoft IIS web server.
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Backup
The UCM630X configuration can be backed up locally or via network. The backup file will be used to restore the configuration on UCM630X when necessary.
Backup/Restore
Users could backup the UCM630X configurations for restore GUIMaintenanceBackupBackup/Restore. Click on "Backup" to create a new backup file. Then the following dialog will show.
purpose
under
Web
Figure 327: Create New Backup
1. Choose the type(s) of files to be included in the backup. 2. Choose where to store the backup file: USB Disk, SD Card, Local, NAS or GDMS. 3. Name the backup file. 4. Click on "Backup" to start backup.
Once the backup is done, the list of the backups will be displayed with date and time in the web page. Users can
download , restore , or delete
it from the UCM630X internal storage or the external device.
Click on
to upload backup file from the local device to UCM630X. The uploaded backup file will also
be displayed in the web page and can be used to restore the UCM630X. Note: users can restore backups of models with more FXO ports to models with less FXO ports as long as the configurations related to the extra FXO ports are removed.
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Please make sure the FXO port settings, total number of extensions and total number of conference rooms are compactable before restoring to another UCM model. Otherwise it will prompt a warning and stop the restore process as shown below:
Figure 328: Restore Warning
Figure 329: Backup / Restore
The
option allows UCM to perform automatically backup on the user specified time.
Regular backup file can only be stored in USB / SD card / SFTP server. User is allowed to set backup time from
0-23 and how frequent the backup will be performed.
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Figure 330: Local Backup
Data Sync
Besides local backup, users could backup the voice records/voice mails/CDR in a daily basis to a remote server via SFTP protocol automatically under Web GUIMaintenanceBackupData Sync. The client account supports special characters such as @ or "." Allowing the use email address as SFTP accounts. It allows users as well to specify the destination directory on SFTP server for backup file. If the directory does not exist on the destination, UCM630X will create the directory automatically
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Figure 331: Data Sync
Enable Data Sync Account Password Server Address Destination Directory
Sync Time
Table 148: Data Sync Configuration
Enable the auto data sync function. The default setting is "No". Enter the Account name on the SFTP backup server. Enter the Password associate with the Account on the SFTP backup server. Enter the SFTP server address. Specify the directory in SFTP server to keep the backup file. Format: `xxx/xxx/xxx', If this directory does not exist, UCM will create this directory automatically. Enter 0-23 to specify the backup hour of the day.
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Before saving the configuration, users could click on
. The UCM630X will then try connecting the
server to make sure the server is up and accessible for the UCM630X. Save the changes and all the backup logs will be listed on the web page. After data sync is configured, users could also manually synchronize all data
by clicking on
instead of waiting for the backup time interval to come.
Restore Configuration from Backup File
To restore the configuration on the UCM630X from a backup file, users could go to Web GUI Maintenance Backup Backup/Restore.
A list of previous configuration backups is displayed on the web page. Users could click on
of the
desired backup file and it will be restored to the UCM630X. If the backup was stored on GDMS, it will be displayed under Backups GDMS Cloud Storage, that can be
restored by clicking on
If users have other backup files on PC to restore on the UCM630X, click on "Upload Backup File" first and select it from local PC to upload on the UCM630X. Once the uploading is done, this backup file will be
displayed in the list of previous configuration backups for restore purpose. Click on backup file.
to restore from the
Figure 332: Restore UCM630X from Backup File
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Note: The uploaded backup file must be a tar file with no special characters like *,!,#,@,&,$,%,^,(,),/,\,space in
the file name. The uploaded back file size must be under 10MB. ---------------------------------------------------------------------------------------------------------------------------------------------------
System Cleanup/Reset
Reset and Reboot
Users could perform reset and reboot under Web GUIMaintenanceSystem Cleanup/ResetReset and Reboot.
To reboot the device, click on reboot icon.
To factory reset the device, click on reset icon, then all the configurations and data will be reset to factory
default.
Figure 333: Reset and Reboot
User can also verify UCM certificate under the same path.
Cleaner
Users could configure to clean the Call Detail Report/Voice Records/Voice Mails etc... manually and automatically under Web GUIMaintenanceSystem Cleanup/ResetCleaner. The following screenshot show the settings and parameters to configure the manual cleaner feature on UCM630X.
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Figure 334: Manual Cleaning
Users can either clean all the data on the UCM or specify the modules to clean such as: Recordings, Fax Files, Voicemail, Music on Hold, Voice Prompts, CDR, ZeroConfig Files, Operation Log, Backups, Coredumps, Troubleshooting, Queue Statistics Reports, Conference Reports, IM Data.
User can also set an automatic cleaning under CleanerAutomatic Cleaning. The following screenshot show the settings and parameters to configure the cleaner feature on UCM630X.
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Enable CDR Cleaner CDR Clean Time Cleaning Conditions
Figure 335: Automatic Cleaning
Table 149: Automatic Cleaning Configuration
Enable the CDR Cleaner function. Enter 0-23 to specify the hour of the day to clean up CDR. By Schedule: If the clean interval is 3, cleaning will be performed every 3 days to remove all records that were generated 3 days ago.
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Keep Last X Records: If the max number of CDR has been reached, CDR will be deleted starting with the oldest entry at the configured cleaning time.(Note: The amount of records displayed on the page of call queue statistics is not one-to-one with the actual amount of records in the database.) Keep Last X Days: Delete all entries older than X days.
Clean Interval
Enter 1-30 to specify the day of the month to clean up CDR when By Schedule is selected as Cleaning Conditions.
Max Entries
Set the maximum number of CDR entries to keep when Keep Last X Records is selected as Cleaning Conditions.
Keep Last X Day
Enter the number of days of call log entries to keep when Keep Last X days is selected as Cleaning Conditions.
Enable Queue Statistics Enable scheduled queue log cleaning. By default, is disabled. Report Cleaner
Queue Statistics Report Enter the hour of the day to start the cleaning. The valid range is 0-23.
Cleaner Clean Time
Cleaning Conditions
By Schedule: If the clean interval is 3, cleaning will be performed every 3 days to remove all records that were generated 3 days ago. Keep Last X Records: If the max number of Queue Statistics has been reached, Queue Statistics will be deleted starting with the oldest entry at the configured cleaning time.(Note: The amount of records displayed on the page of call queue statistics is not one-to-one with the actual amount of records in the database.) Keep Last X Days: Delete all entries older than X days.
Clean Interval
Enter how often (in days) to clean queue logs when By Schedule is selected as Cleaning Conditions. The valid range is 1-30.
Max Entries
Set the maximum number of Queue Statistics entries to keep when Keep Last X Records is selected as Cleaning Conditions.
Keep Last X Day
Enter the number of days of call log entries to keep when Keep Last X days is selected as Cleaning Conditions.
Enable Conference Statistics Report Cleaner
Enable scheduled Conference log cleaning. By default, is disabled.
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Cleaning Conditions
Clean Interval Max Entries Keep Last X Day Enable File Cleaner Clean Files in External Device Choose Cleaner File
Clean time Cleaning Conditions
File Clean Interval File Clean Threshold
By Schedule: If the clean interval is 3, cleaning will be performed every 3 days to remove all records that were generated 3 days ago.
Keep Last X Records: If the max number of Conference Statistics Report has been reached, Conference Statistics Report will be deleted starting with the oldest entry at the configured cleaning time. (Note: The amount of records displayed on the page of call queue statistics is not one-to-one with the actual amount of records in the database.)
Keep Last X Days: Delete all entries older than X days. Enter how often (in days) to clean queue logs when By Schedule is selected as Cleaning Conditions. The valid range is 1-30. Set the maximum number of Conference Statistics Report entries to keep when Keep Last X Records is selected as Cleaning Conditions. Enter the number of days of call log entries to keep when Keep Last X days is selected as Cleaning Conditions. Enter the Voice Records Cleaner function. If enabled the files in external device (USB/SD card) will be atomically cleaned up as configured. Select the files for system automatic clean. Basic Call Recording Files. Conference Recording Files. Call Queue Recording Files. Voicemail Files. Backup Files. Enter the hour of the day to start the cleaning. The valid range is 0-23.
By Schedule: If the clean interval is 3, cleaning will be performed every 3 days to delete all files.
By Threshold: Check at the configured cleaning time every day to see if the storage threshold has been exceeded and perform cleaning of all files if it has.
Keep Last X Days: Delete all files older than X days. Enter 1-30 to specify the day of the month to clean up the files. Enter the internal storage disk usage threshold (in percent). Once this threshold is exceeded, the file cleanup will proceed as scheduled. Valid
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Keep Last X Days Cleaner Log
range is 0-99. Automatically delete all recordings older than this x days when the threshold is reached. If not set, all data is cleared Press Clean "button" to clean cleaner log.
All the cleaner logs will be listed on the bottom of the page.
USB/SD Card Files Cleanup
Users could configure to clean or download the Call Detail Report/Voice Records/Voice Mails automatically under Web GUIMaintenanceSystem Cleanup/ResetUSB / SD Card Files Cleanup.
Figure 336: USB/SD Card Files Cleanup
Current Path Directory Delete Selected File
Table 150: USB/SD Card Files Cleanup
Displays the current path. Select the directory user want to clean. Select multiple entries to delete from USB or SD card.
System Recovery
In some cases (for example after wrong upgrading procedure where the user doesn't follow the correct steps to perform an upgrade) the system may go into some hardware/software issues where the web UI access is lost
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as well as SSH, in this case the only solution would be to perform a full system recovery in order to reset or update the software version of the device in order to use it again.
1. To access recovery mode on UCM, please follow below steps: 2. Remove the power from the unit and keep the network cable connected. 3. Press using a PIN the reset button and keep holding. 4. Plug back the power supply while maintaining the reset button pressed. 5. Wait for couple of seconds until you hear a click sound. 6. Release the reset button, and the system should display on the LCD a message "Recovery Mode" along
with an IP address.
Once at this stage, the administrator can access the recovery mode web portal by typing in either the IP0 address (typically WAN) or IP1 address (typically LAN) into a browser address bar. The following page should appear:
Figure 337: UCM6302 Recovery Web Page
Make sure to enter the correct admin password, and press login to access the recovery mode page :
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Figure 338: Recovery Mode
From here, the user can either upload a firmware file, factory reset or just reboot the device.
Syslog
On the UCM630X, users could dump the syslog information to a remote server under Web GUIMaintenanceSyslog. Enter the syslog server hostname or IP address and select the module/level for the syslog information as well as Process Log Level. The default syslog level for all modules is "error", which is recommended in your UCM630X settings because it can be helpful to locate the issues when errors happen. Some typical modules for UCM630X functions are as follows and users can turn on "NOTICE" and "VERBOSE" levels besides "error" level.
pbx: This module is related to general PBX functions. pjsip: This module is related to SIP calls. chan_dahdi: This module is related to analog calls (FXO/FXS).
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Note: Syslog is usually for debugging and troubleshooting purpose. Turning on all levels for all syslog modules is not recommended for daily usage. Too many syslog prints might cause traffic and affect system performance. The reserved size for Syslog entries on the cache memory of the UCM is 50M, once this sized is reached the
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UCM will clean up 2M of the oldest Syslog entries to allow to save new logs. ---------------------------------------------------------------------------------------------------------------------------------------------------
Network Troubleshooting
On the UCM630X, users could capture traces, ping remote host and traceroute remote host for troubleshooting purpose under Web GUIMaintenanceNetwork Troubleshooting. The following sections shows the steps to capture different types of traffic traces for analysis purposes. Ethernet Capture The captured trace can be downloaded for analysis. The instructions or result will be displayed in the Web GUI output result.
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Capture Type
Interface Type Enable SFTP Data Sync
Figure 339: Ethernet Capture
Table 151: Ethernet Capture
Ethernet Capture: Gets a packet capture of all network traffic going through the device. WebSocket Capture: Gets a packet capture of WebSocket protocol. Mainly used for troubleshooting GS Wave Web calling and conferencing issues. Select the network interface to monitor. Check this box to save the capture files in the SFTP server. Please make sure the configuration of data synchronization works before.
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Storage to External Device
Capture Filter
Save to external storage Start Stop Download Enable SRTP Debugging
Check this box to activate storage of the capture either on the USB or SD Card. Enter the filter to obtain the specific types of traffic, such as (host, src, dst, net, proto...). Save to external storage Click to start the trace. Click to stop the trace. Click to download the trace if trace is stored locally. Check this box to troubleshoot calls encrypted with TLS/SRTP.
The output result is in .pcap format. Therefore, users could specify the capture filter as used in general network traffic capture tool (host, src, dst, net, protocol, port, port range) before starting to capture the trace. Note: Capture files saved on external devices will now have "capture" prepended to file names.
IP Ping
Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below.
Figure 340: Ping
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Traceroute
Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below.
Figure 341: Traceroute
Signaling Troubleshooting Analog Record Trace
Analog Record Trace
Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM630X user has caller ID issue for incoming call from Analog trunk. Users can access analog record trance under Web GUIMaintenanceSignal Troubleshooting.
Here is the step to capture trace:
1. Select FXO or FXS for "Record Ports". If the issue happens on FXO 1, select FXO port 1 to record the trace.
2. Click on "Start". 3. Make a call via the analog port that has the issue. 4. Once done, click on "Stop".
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5. Click on "Download" to download the analog record trace.
Figure 342: Troubleshooting Analog Trunks
A key Dial-up FXO
Users can directly set a PSTN number on the "External Extension" text box to troubleshoot issues related to the analog trunk easily, the following steps shows how to use this feature:
1. Configure analog trunk on UCM, including outbound route. 2. Enter a reachable external number in "External Extension". 3. Press "Start" button. The call will be initiated to the external number. 4. Answer and finish the call before pressing "Stop" button. The trace will be available for analysis to download after output result shows "Done! Click on Download to download the captured packets".
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Figure 343: A Key Dial-up FXO
Note: When using a Key Dial-up FXO feature the outbound trunk for the analog trunk need to have internal permission. As well as it should be the trunk with the highest outbound route priority.
After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream Technical support in the following link for further assistance if the issue is not resolved. http://www.grandstream.com/index.php/support
Service Check
Enable Service Check to periodically check UCM630X. Check Cycle is configurable in seconds and the default setting is 60 sec. Check Times is the maximum number of failed checks before restart the UCM630X. The default setting is 3. If there is no response from UCM630X after 3 attempts (default) to check, current status will be stored and the internal service in UCM630X will be restarted.
Figure 344: Service Check
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Network Status
In UCM630X Web GUISystem StatusNetwork Status, the users can view active Internet connections. This information can be used to troubleshoot connection issue between UCM630X and other services.
Figure 345: Network Status
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EXPERIENCING THE UCM630X SERIES IP PBX
Please visit our website: http://www.grandstream.com to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products. We encourage you to browse our product related documentation, FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream Certified Partner or Reseller, please contact them directly for immediate support. Our technical support staff is trained and ready to answer all of your questions. Contact a technical support member or submit a trouble ticket online to receive in-depth support. Thank you again for purchasing Grandstream UCM630X series IP PBX appliance, it will be sure to bring convenience and color to both your business and personal life.
* Asterisk is a Registered Trademark of Digium, Inc
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