Grandstream Networks UCM6202 IP PBX User Manual
Grandstream Networks, Inc. IP PBX
User Manual
Grandstream Networks, Inc.
UCM6200 Series IP PBX
User Manual
Firmware Version 1.0.0.1
UCM6200 Series IP PBX User Manual
Page 1 of 321
UCM6200 Series IP PBX User Manual
Table of Content
GNU GPL INFORMATION ........................................................................ 17
CHANGE LOG ......................................................................................... 18
FIRMWARE VERSION 1.0.0.1 ............................................................................................................ 18
WELCOME ............................................................................................... 19
PRODUCT OVERVIEW ............................................................................ 21
TECHNICAL SPECIFICATIONS ......................................................................................................... 21
INSTALLATION ........................................................................................ 25
EQUIPMENT PACKAGING ................................................................................................................. 25
CONNECT YOUR UCM6200 .............................................................................................................. 25
CONNECT THE UCM6202 ........................................................................................................... 25
CONNECT THE UCM6204 ........................................................................................................... 26
CONNECT THE UCM6208 ........................................................................................................... 27
SAFETY COMPLIANCES ................................................................................................................... 28
WARRANTY ........................................................................................................................................ 28
GETTING STARTED ................................................................................ 31
USE THE LCD MENU ......................................................................................................................... 31
USE THE LED INDICATORS .............................................................................................................. 33
USE THE WEB GUI ............................................................................................................................ 34
ACCESS WEB GUI ...................................................................................................................... 34
SETUP WIZARD ........................................................................................................................... 36
WEB GUI CONFIGURATIONS ..................................................................................................... 37
WEB GUI LANGUAGES ............................................................................................................... 37
SAVE AND APPLY CHANGES ..................................................................................................... 38
MAKE YOUR FIRST CALL .................................................................................................................. 38
SYSTEM SETTINGS ................................................................................ 39
USER MANAGEMENT ........................................................................................................................ 39
USER PRIVILEGES ..................................................................................................................... 39
CREATE NEW WEB UI USER ..................................................................................................... 40
USER PORTAL ............................................................................................................................. 42
CONCURRENT MULTI-USER LOGIN ......................................................................................... 43
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OPERATION LOG ........................................................................................................................ 43
CHANGE PASSWORD ....................................................................................................................... 45
CHANGE BINDNG EMAIL ........................................................................................................... 46
NETWORK SETTINGS ....................................................................................................................... 47
BASIC SETTINGS ........................................................................................................................ 47
802.1X .......................................................................................................................................... 52
STATIC ROUTES.......................................................................................................................... 53
PORT FORWORDING.................................................................................................................. 56
DDNS SETTINGS ......................................................................................................................... 58
FIREWALL ........................................................................................................................................... 60
STATIC DEFENSE ....................................................................................................................... 60
DYNAMIC DEFENSE ................................................................................................................... 63
FAIL2BAN ..................................................................................................................................... 64
LDAP SERVER .................................................................................................................................... 65
LDAP SERVER CONFIGURATIONS ........................................................................................... 66
LDAP PHONEBOOK .................................................................................................................... 67
LDAP CLIENT CONFIGURATIONS ............................................................................................. 70
HTTP SERVER .................................................................................................................................... 72
EMAIL SETTINGS ............................................................................................................................... 73
TIME SETTINGS ................................................................................................................................. 74
AUTO TIME UPDATING ............................................................................................................... 74
SET TIME MANUALLY ................................................................................................................. 76
OFFICE TIME ............................................................................................................................... 76
HOLIDAY ...................................................................................................................................... 78
NTP SERVER ...................................................................................................................................... 80
RECORDINGS STORAGE .................................................................................................................. 80
LOGIN TIMEOUT SETTINGS ............................................................................................................. 82
GOOGLE SERVICE SETTINGS SUPPORT ....................................................................................... 83
PROVISIONING ....................................................................................... 87
OVERVIEW ......................................................................................................................................... 87
CONFIGURATION ARCHITECTURE FOR END POINT DEVICE ...................................................... 87
AUTO PROVISIONING SETTINGS .................................................................................................... 88
DISCOVERY ........................................................................................................................................ 91
GLOBAL CONFIGURATION ............................................................................................................... 92
GLOBAL POLICY ......................................................................................................................... 92
GLOBAL TEMPLATES ............................................................................................................... 100
MODEL CONFIGURATION ............................................................................................................... 102
MODEL TEMPLATES ................................................................................................................. 102
MODEL UPDATE ........................................................................................................................ 104
DEVICE CONFIGURATION .............................................................................................................. 105
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CREATE NEW DEVICE .............................................................................................................. 105
MANAGE DEVICES ................................................................................................................... 106
SAMPLE APPLICATION .................................................................................................................... 113
EXTENSIONS ......................................................................................... 119
CREATE NEW USER ........................................................................................................................ 119
CREATE NEW SIP EXTENSION ............................................................................................... 119
CREATE NEW IAX EXTENSION ............................................................................................... 125
CREATE NEW FXS EXTENSION .............................................................................................. 129
BATCH ADD EXTENSIONS .............................................................................................................. 134
BATCH ADD SIP EXTENSIONS ................................................................................................ 134
BATCH ADD IAX EXTENSIONS ................................................................................................ 137
SEARCH AND EDIT EXTENSION .................................................................................................... 139
EXPORT EXTENSIONS .................................................................................................................... 141
IMPORT EXTENSIONS..................................................................................................................... 141
EMAIL TO USER ............................................................................................................................... 142
MULTIPLE REGISTRATIONS PER EXTENSION ............................................................................. 144
SMS MESSAGE SUPPORT .............................................................................................................. 145
TRUNKS ................................................................................................. 147
ANALOG TRUNKS ............................................................................................................................ 147
ANALOG TRUNK CONFIGURATION ........................................................................................ 147
PSTN DETECTION .................................................................................................................... 150
VOIP TRUNKS .................................................................................................................................. 153
DIRECT OUTWARD DIALING (DOD) ........................................................................................ 164
SLA STATION ........................................................................................ 167
CREATE/EDIT SLA STATION ........................................................................................................... 167
SAMPLE CONFIGURATION ............................................................................................................. 168
CALL ROUTES ...................................................................................... 171
OUTBOUND ROUTES ...................................................................................................................... 171
INBOUND ROUTES .......................................................................................................................... 173
INBOUND RULE CONFIGURATIONS ....................................................................................... 174
INBOUND ROUTE: PREPEND EXAMPLE ................................................................................ 175
INBOUND ROUTE: MULTIPLE MODE ...................................................................................... 176
FAX INTELLIGENT ROUTE ....................................................................................................... 177
FAX WITH TWO MEDIA ............................................................................................................. 178
BLACKLIST CONFIGURATIONS ............................................................................................... 178
CONFERENCE BRIDGE ........................................................................ 181
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CONFERENCE BRIDGE CONFIGURATIONS .......................................................................... 181
JOIN A CONFERENCE CALL .................................................................................................... 183
INVITE OTHER PARTIES TO JOIN CONFERENCE ................................................................. 183
DURING THE CONFERENCE ................................................................................................... 184
RECORD CONFERENCE .......................................................................................................... 185
CONFERENCE SCHEDULE .................................................................. 187
CONFERENCE SCHEUDLE CONFIGURATION .............................................................................. 187
IVR ......................................................................................................... 191
CONFIGURE IVR .............................................................................................................................. 191
CREATE CUSTOM PROMPT ........................................................................................................... 193
RECORD NEW CUSTOM PROMPT .......................................................................................... 193
UPLOAD CUSTOM PROMPT .................................................................................................... 194
LANGUAGE SETTINGS FOR VOICE PROMPT .................................... 195
DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE ............................................................... 195
CUSTOMIZE SPECIFIC PROMPT ................................................................................................... 197
VOICEMAIL ............................................................................................ 199
CONFIGURE VOICEMAIL ................................................................................................................ 199
ACCESS VOICEMAIL ....................................................................................................................... 200
VOICEMAIL EMAIL SETTINGS ........................................................................................................ 201
CONFIGURE VOICEMAIL GROUP .................................................................................................. 202
RING GROUP ......................................................................................... 205
CONFIGURE RING GROUP ............................................................................................................. 205
REMOTE EXTENSION IN RING GROUP ........................................................................................ 207
PAGING AND INTERCOM GROUP ....................................................... 211
CONFIGURE PAGING/INTERCOM GROUP .................................................................................... 211
CALL QUEUE ........................................................................................ 213
CONFIGURE CALL QUEUE ............................................................................................................. 213
EXTENSION GROUPS ........................................................................... 217
CONFIGURE EXTENSION GROUPS .............................................................................................. 217
USING EXTENSION GROUPS ......................................................................................................... 218
PICKUP GROUPS .................................................................................. 219
CONFIGURE PICKUP GROUPS ...................................................................................................... 219
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CONFIGURE PICKUP FEATURE CODE.......................................................................................... 219
MUSIC ON HOLD ................................................................................... 221
FAX/T.38 ................................................................................................. 223
CONFIGURE FAX/T.38 ..................................................................................................................... 223
SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE ............................................... 224
SAMPLE CONFIGURATION FOR FAX-TO-EMAIL .......................................................................... 227
ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) ............ 229
BUSY CAMP-ON .................................................................................... 231
FOLLOW ME .......................................................................................... 233
ONE-KEY DIAL ...................................................................................... 237
DISA ....................................................................................................... 239
CALLBACK FEATURE .......................................................................... 241
BLF AND EVENT LIST ........................................................................... 243
BLF .................................................................................................................................................... 243
EVENT LIST ...................................................................................................................................... 243
DIAL BY NAME ...................................................................................... 247
DIAL BY NAME CONFIGURATION................................................................................................... 247
ACTIVE CALLS AND MONITOR ........................................................... 251
ACTIVE CALLS STATUS .................................................................................................................. 251
HANG UP ACTIVE CALLS ................................................................................................................ 252
CALL MONITOR ................................................................................................................................ 252
CALL FEATURES .................................................................................. 255
FEATURE CODES ............................................................................................................................ 255
CALL RECORDING ........................................................................................................................... 259
CALL PARK ....................................................................................................................................... 260
PARK A CALL ............................................................................................................................. 260
RETRIEVE THE PARKED CALL ................................................................................................ 260
ENABLE SPY .................................................................................................................................... 260
INTERNAL OPTIONS ............................................................................. 263
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INTERNAL OPTIONS/GENERAL ..................................................................................................... 263
INTERNAL OPTIONS/JITTER BUFFER ........................................................................................... 265
INTERNAL OPTIONS/RTP SETTINGS ............................................................................................ 266
INTERNAL OPTIONS/PAYLOAD ...................................................................................................... 266
IAX SETTINGS ....................................................................................... 269
IAX SETTINGS/GENERAL................................................................................................................ 269
IAX SETTINGS/REGISTRATION ...................................................................................................... 269
IAX SETTINGS/STATIC DEFENSE .................................................................................................. 270
SIP SETTINGS ....................................................................................... 271
SIP SETTINGS/GENERAL ................................................................................................................ 271
SIP SETTINGS/MISC ........................................................................................................................ 271
SIP SETTINGS/SESSION TIMER ..................................................................................................... 272
SIP SETTINGS/TCP AND TLS .......................................................................................................... 272
SIP SETTINGS/NAT .......................................................................................................................... 273
SIP SETTINGS/TOS.......................................................................................................................... 274
PORTS CONFIG .................................................................................... 277
VALUE-ADDED FEATURES .................................................................. 279
FAX SENDING .................................................................................................................................. 279
ANNOUNCEMENTS CENTER ......................................................................................................... 279
ANNOUNCEMENTS CENTER SETTINGS ............................................................................... 280
GROUP SETTINGS .................................................................................................................... 280
STATUS AND REPORTING ................................................................... 283
PBX STATUS ..................................................................................................................................... 283
TRUNKS ..................................................................................................................................... 283
EXTENSIONS ............................................................................................................................. 284
QUEUES ..................................................................................................................................... 285
CONFERENCE ROOMS ............................................................................................................ 286
INTERFACES STATUS............................................................................................................... 287
PARKING LOT ............................................................................................................................ 288
SYSTEM STATUS ............................................................................................................................. 289
GENERAL ................................................................................................................................... 289
NETWORK ................................................................................................................................. 290
STORAGE USAGE ..................................................................................................................... 290
RESOURCE USAGE .................................................................................................................. 291
SYSTEM EVENTS ............................................................................................................................ 292
ALERT EVENTS LIST ................................................................................................................ 292
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ALERT LOG ................................................................................................................................ 294
ALERT CONTACT ...................................................................................................................... 295
CDR ................................................................................................................................................... 295
CDR IMPROVEMENT ................................................................................................................ 298
DOWNLOADED CDR FILE ........................................................................................................ 299
STATISTICS ................................................................................................................................ 301
RECORDING FILES ................................................................................................................... 301
API CONFIGURATION ............................................................................................................... 302
UPGRADING AND MAINTENANCE ...................................................... 305
UPGRADING ..................................................................................................................................... 305
UPGRADING VIA NETWORK .................................................................................................... 305
UPGRADING VIA LOCAL UPLOAD ........................................................................................... 306
NO LOCAL FIRMWARE SERVERS ........................................................................................... 308
BACKUP ............................................................................................................................................ 308
BACKUP/RESTORE ................................................................................................................... 308
DATA SYNC ................................................................................................................................ 311
RESTORE CONFIGURATION FROM BACKUP FILE ............................................................... 312
CLEANER .......................................................................................................................................... 313
RESET AND REBOOT ...................................................................................................................... 314
SYSLOG ............................................................................................................................................ 315
TROUBLESHOOTING ...................................................................................................................... 315
ETHERNET CAPTURE .............................................................................................................. 315
IP PING ....................................................................................................................................... 316
TRACEROUTE ........................................................................................................................... 317
ANALOG RECORD TRACE ....................................................................................................... 317
SERVICE CHECK ...................................................................................................................... 318
NETWORK STATUS ................................................................................................................... 318
REMOTE ACCESS ............................................................................................................................ 319
SSH ACCESS ............................................................................................................................. 319
EXPERIENCING THE UCM6200 SERIES IP PBX ................................. 321
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Table of Tables
UCM6200 Series IP PBX User Manual
Table 1: Technical Specifications................................................................................................................. 21
Table 2: UCM6202/UCM6204 Equipment Packaging ................................................................................. 25
Table 3: LCD Menu Options ........................................................................................................................ 32
Table 4: UCM6202/UCM6204 LED INDICATORS ...................................................................................... 33
Table 5: UCM6208 LED INDICATORS ....................................................................................................... 34
Table 6: User Management->Create New User .......................................................................................... 41
Table 7: Operation Log Column Header ..................................................................................................... 44
Table 8: Change Binding Email option ........................................................................................................ 47
Table 9: UCM6202/UCM6204 Network Settings->Basic Settings .............................................................. 47
Table 10: UCM6208 Network Settings->Basic Settings .............................................................................. 49
Table 11: UCM6200 Network Settings->802.1X ......................................................................................... 53
Table 12: UCM6200 Network Settings->Static Routes ............................................................................... 54
Table 13: UCM6202/UCM6204 Network Settings->Port Forwarding .......................................................... 56
Table 14: UCM6200 Firewall->Static Defense->Current Service ................................................................ 61
Table 15: Typical Firewall Settings .............................................................................................................. 61
Table 16: Firewall Rule Settings .................................................................................................................. 62
Table 17: UCM6200 Firewall Dynamic Defense ......................................................................................... 63
Table 18: Fail2Ban Settings ........................................................................................................................ 64
Table 19: HTTP Server Settings .................................................................................................................. 73
Table 20: Email Settings .............................................................................................................................. 73
Table 21: Time Auto Updating ..................................................................................................................... 75
Table 22: Create New Office Time .............................................................................................................. 77
Table 23: Create New Holiday..................................................................................................................... 79
Table 24: Auto Provision Settings ............................................................................................................... 90
Table 25: Global Policy Parameters->Localization ..................................................................................... 93
Table 26: Global Policy Parameters->Phone Settings ................................................................................ 94
Table 27: Global Policy Parameters->Contact List ..................................................................................... 95
Table 28: Global Policy Parameters->Maintenance .................................................................................... 97
Table 29: Global Policy Parameters->Network Settings ............................................................................. 98
Table 30: Global Policy Parameters->Customization.................................................................................. 99
Table 31: Create New Template ................................................................................................................ 100
Table 32: Create New Model Template ..................................................................................................... 102
Table 33: SIP Extension Configuration Parameters->Basic Settings ....................................................... 120
Table 34: SIP Extension Configuration Parameters->Media..................................................................... 121
Table 35: SIP Extension Configuration Parameters->Features ................................................................ 122
Table 36: SIP Extension Configuration Parameters->Specific Time ......................................................... 125
Table 37: IAX Extension Configuration Parameters->Basic Settings ....................................................... 125
Table 38: IAX Extension Configuration Parameters->Media .................................................................... 126
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Table 39: IAX Extension Configuration Parameters->Features ................................................................ 127
Table 40: IAX Extension Configuration Parameters->Specific Time ......................................................... 129
Table 41: FXS Extension Configuration Parameters->Basic Settings ...................................................... 130
Table 42: FXS Extension Configuration Parameters->Media ................................................................... 131
Table 43: FXS Extension Configuration Parameters->Features ............................................................... 132
Table 44: FXS Extension Configuration Parameters->Specific Time ........................................................ 134
Table 45: Batch Add SIP Extension Parameters ....................................................................................... 135
Table 46: Batch Add IAX Extension Parameters ....................................................................................... 137
Table 47: Analog Trunk Configuration Parameters ................................................................................... 147
Table 48: PSTN Detection for Analog Trunk ............................................................................................. 152
Table 49: Create New SIP Trunk ............................................................................................................... 154
Table 50: SIP Register Trunk Configuration Parameters .......................................................................... 155
Table 51: SIP Peer Trunk Configuration Parameters ................................................................................ 158
Table 52: Create New IAX Trunk ............................................................................................................... 161
Table 53: IAX Register Trunk Configuration Parameters .......................................................................... 161
Table 54: IAX Peer Trunk Configuration Parameters ................................................................................ 162
Table 55: SLA Station Configuration Parameters ...................................................................................... 167
Table 56: Outbound Route Configuration Parameters .............................................................................. 171
Table 57: Inbound Rule Configuration Parameters ................................................................................... 174
Table 58: Conference Bridge Configuration Parameters .......................................................................... 181
Table 59: Conference Caller IVR Menu .................................................................................................... 185
Table 60: Conference Schedule Parameters ............................................................................................ 187
Table 61: IVR Configuration Parameters .................................................................................................. 191
Table 62: Voicemail Settings ..................................................................................................................... 199
Table 63: Voicemail IVR Menu .................................................................................................................. 200
Table 64: Voicemail Email Settings ........................................................................................................... 201
Table 65: Voicemail Group Settings .......................................................................................................... 203
Table 66: Ring Group Parameters ............................................................................................................ 205
Table 67: Paging/Intercom Group Configuration Parameters ................................................................... 211
Table 68: Call Queue Configuration Parameters ...................................................................................... 213
Table 69: FAX/T.38 Settings ...................................................................................................................... 223
Table 70: Follow Me Settings .................................................................................................................... 234
Table 71: Follow Me Options ..................................................................................................................... 235
Table 72: DISA Settings ............................................................................................................................ 239
Table 73: Callback Configuration Parameters........................................................................................... 241
Table 74: Event List Settings ..................................................................................................................... 243
Table 75: UCM6200 Feature Codes ......................................................................................................... 255
Table 76: Internal Options/General ........................................................................................................... 263
Table 77: Internal Options/Jitter Buffer ...................................................................................................... 265
Table 78: Internal Options/RTP Settings ................................................................................................... 266
Table 79: Internal Options/Payload ........................................................................................................... 266
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Table 80: IAX Settings/General ................................................................................................................. 269
Table 81: IAX Settings/Registration .......................................................................................................... 269
Table 82: IAX Settings/Static Defense ...................................................................................................... 270
Table 83: SIP Settings/General ................................................................................................................. 271
Table 84: SIP Settings/Misc ...................................................................................................................... 271
Table 85: SIP Settings/Session Timer ....................................................................................................... 272
Table 86: SIP Settings/TCP and TLS ........................................................................................................ 272
Table 87: SIP Settings/NAT ....................................................................................................................... 273
Table 88: SIP Settings/ToS ........................................................................................................................ 274
Table 89: Internal Options/Ports Config .................................................................................................... 277
Table 90: Announcements Center Settings ............................................................................................... 280
Table 91: Group Settings ........................................................................................................................... 280
Table 92: Trunk Status ............................................................................................................................... 283
Table 93: Extension Status ........................................................................................................................ 285
Table 94: Agent Status .............................................................................................................................. 286
Table 95: Interface Status Indicators ......................................................................................................... 287
Table 96: Parking Lot Status ..................................................................................................................... 288
Table 97: System Status->General ........................................................................................................... 289
Table 98: System Status->Network ........................................................................................................... 290
Table 99: CDR Filter Criteria ..................................................................................................................... 296
Table 100: CDR Statistics Filter Criteria .................................................................................................... 301
Table 101: API Configuration Files ............................................................................................................ 302
Table 102: Network Upgrade Configuration .............................................................................................. 306
Table 103: Data Sync Configuration ......................................................................................................... 311
Table 104: Cleaner Configuration ............................................................................................................. 313
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Table of Figures
UCM6200 Series IP PBX User Manual
Figure 1: UCM6202 Front View ................................................................................................................... 25
Figure 2: UCM6202 Back View ................................................................................................................... 26
Figure 3: UCM6204 Front View ................................................................................................................... 26
Figure 4: UCM6204 Back View ................................................................................................................... 27
Figure 5: UCM6208 Front View ................................................................................................................... 28
Figure 6: UCM6208 Back View ................................................................................................................... 28
Figure 7: UCM6204 Web GUI Login Page .................................................................................................. 34
Figure 8: UCM6200 Setup Wizard .............................................................................................................. 36
Figure 9: UCM6200 Web GUI Language .................................................................................................... 37
Figure 10: User Management Page Display ............................................................................................... 39
Figure 11: Create New User ........................................................................................................................ 41
Figure 12: User Management – New Users ................................................................................................ 41
Figure 13: Edit User Information by Super Admin....................................................................................... 42
Figure 14: User Portal Login ....................................................................................................................... 42
Figure 15: User Portal Layout ..................................................................................................................... 43
Figure 16: Multiple User Operation Error Prompt ....................................................................................... 43
Figure 17: Operation Logs .......................................................................................................................... 44
Figure 18: Operation Logs Filter ................................................................................................................. 45
Figure 19 : Change Password..................................................................................................................... 46
Figure 20: Change Binding Email ............................................................................................................... 46
Figure 21: UCM6200 Network Interface Method: Route ............................................................................. 50
Figure 22: UCM6200 Network Interface Method: Switch ............................................................................ 51
Figure 23: UCM6200 Network Interface Method: Dual ............................................................................... 52
Figure 24: UCM6200 Using 802.1X as Client ............................................................................................. 52
Figure 25: UCM6200 Using 802.1X EAP-MD5 ........................................................................................... 53
Figure 26: UCM6204 Static Route Sample ................................................................................................. 55
Figure 27: UCM6204 Static Route Configuration ........................................................................................ 55
Figure 28: UCM6202/UCM6204 Port Forwarding Configuration ................................................................ 57
Figure 29: GXP2160 Web Access Using UCM6202 Port Forwarding ........................................................ 58
Figure 30: Register Domain Name on noip.com ......................................................................................... 59
Figure 31: UCM6200 DDNS Setting ........................................................................................................... 59
Figure 32: Using Domain Name to Connect to UCM6200 .......................................................................... 60
Figure 33: Create New Firewall Rule .......................................................................................................... 62
Figure 34: Configure Dynamic Defense ...................................................................................................... 64
Figure 35: LDAP Server Configurations ...................................................................................................... 66
Figure 36: Default LDAP Phonebook DN .................................................................................................... 66
Figure 37: Default LDAP Phonebook Attributes .......................................................................................... 67
Figure 38: LDAP Server->LDAP Phonebook .............................................................................................. 67
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Figure 39: Add LDAP Phonebook ............................................................................................................... 68
Figure 40: Edit LDAP Phonebook ............................................................................................................... 68
Figure 41: Import Phonebook...................................................................................................................... 69
Figure 42: Phonebook CSV File Format ..................................................................................................... 69
Figure 43: LDAP Phonebook After Import ................................................................................................... 70
Figure 44: Export Selected LDAP Phonebook ............................................................................................ 70
Figure 45: LDAP Client Configurations ....................................................................................................... 71
Figure 46: GXP2200 LDAP Phonebook Configuration ............................................................................... 72
Figure 47: UCM6200 Email Settings ........................................................................................................... 74
Figure 48: Set Time Manually ..................................................................................................................... 76
Figure 49: Create New Office Time ............................................................................................................. 77
Figure 50: Settings->Time Settings->Office Time ....................................................................................... 78
Figure 51: Create New Holiday ................................................................................................................... 78
Figure 52: Settings->Time Settings->Holiday ............................................................................................. 79
Figure 53: Settings->Recordings Storage ................................................................................................... 80
Figure 54: Recordings Storage Prompt Information ................................................................................... 81
Figure 55: Recording Storage Category ..................................................................................................... 81
Figure 56: Login Timeout Settings .............................................................................................................. 82
Figure 57: Google Service Settings->OAuth2.0 Authentication .................................................................. 83
Figure 58: Google Service->New Project ................................................................................................... 84
Figure 59: Google Service->Create New Credential .................................................................................. 84
Figure 60: Google Service->OAuth2.0 Login .............................................................................................. 85
Figure 61: Zero Config Configuration Architecture for End Point Device ................................................... 88
Figure 62: UCM6200 Zero Config ............................................................................................................... 89
Figure 63: Auto Provision Settings .............................................................................................................. 90
Figure 64: Auto Discover ............................................................................................................................. 92
Figure 65: Discovered Devices ................................................................................................................... 92
Figure 66: Global Policy Categories ........................................................................................................... 93
Figure 67: Edit Global Template ................................................................................................................ 101
Figure 68: Edit Model Template ................................................................................................................ 103
Figure 69: Template Management ............................................................................................................ 104
Figure 70: Upload Model Template Manually ............................................................................................ 105
Figure 71: Create New Device .................................................................................................................. 106
Figure 72: Manage Devices ...................................................................................................................... 106
Figure 73: Edit Device ............................................................................................................................... 107
Figure 74: Edit Customize Device Settings ............................................................................................... 109
Figure 75: Add P Value in Customize Device Settings ............................................................................. 110
Figure 76: Modify Selected Devices - Same Model .................................................................................. 111
Figure 77: Modify Selected Devices - Different Models ............................................................................ 112
Figure 78: Device List in Zero Config ........................................................................................................ 113
Figure 79: Zero Config Sample - Global Policy......................................................................................... 114
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Figure 80: Zero Config Sample - Device Preview 1 .................................................................................. 115
Figure 81: Zero Config Sample - Device Preview 2 .................................................................................. 116
Figure 82: Zero Config Sample - Device Preview 3 .................................................................................. 117
Figure 83: Create New Device .................................................................................................................. 119
Figure 84: Manage Extensions ................................................................................................................. 140
Figure 85: Export Extensions .................................................................................................................... 141
Figure 86: Import Extensions .................................................................................................................... 141
Figure 87: Email To User - Prompt Information ......................................................................................... 142
Figure 88: Account Registration Information and QR Code ...................................................................... 143
Figure 89: LDAP Client Information and QR Code ................................................................................... 143
Figure 90: Multiple Registrations per Extension ....................................................................................... 144
Figure 91: Extension - Concurrent Registration ........................................................................................ 144
Figure 92: SMS Message Support ............................................................................................................ 145
Figure 93: UCM6200 FXO Tone Settings ................................................................................................. 150
Figure 94: UCM6200 PSTN Detection ...................................................................................................... 151
Figure 95: UCM6200 PSTN Detection: Auto Detect ................................................................................. 151
Figure 96: UCM6200 PSTN Detection: Semi-Auto Detect ....................................................................... 152
Figure 97: DOD extension selection ......................................................................................................... 164
Figure 98: Edit DOD .................................................................................................................................. 165
Figure 99: SLA Station .............................................................................................................................. 167
Figure 100: Enable SLA Mode for Analog Trunk ....................................................................................... 169
Figure 101: Analog Trunk with SLA Mode Enabled .................................................................................. 169
Figure 102: SLA Example - SLA Station ................................................................................................... 169
Figure 103: SLA Example - MPK Configuration ........................................................................................ 170
Figure 104: Inbound Route feature: Prepend ........................................................................................... 176
Figure 105: Inbound Route - Multiple Mode.............................................................................................. 177
Figure 106: Blacklist Configuration Parameters........................................................................................ 178
Figure 107: Blacklist csv File..................................................................................................................... 179
Figure 108: Conference Invitation From Web GUI .................................................................................... 183
Figure 109: Conference Recording ........................................................................................................... 186
Figure 110: Conference Schedule............................................................................................................. 190
Figure 111: Click on Prompt to Create IVR Prompt .................................................................................. 193
Figure 112: Record New Custom Prompt ................................................................................................. 194
Figure 113: Upload Custom Prompt .......................................................................................................... 194
Figure 114: Language Settings for Voice Prompt ..................................................................................... 195
Figure 115: Voice Prompt Package List .................................................................................................... 196
Figure 116: New Voice Prompt Language Added ..................................................................................... 196
Figure 117: Upload Single Voice Prompt for Entire Language Pack ........................................................ 197
Figure 118: Voicemail Email Settings ........................................................................................................ 202
Figure 119: Voicemail Group ..................................................................................................................... 203
Figure 120: Ring Group ............................................................................................................................. 205
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Figure 121: Ring Group Configuration ...................................................................................................... 207
Figure 122: Sync LDAP Server option ...................................................................................................... 208
Figure 123: Manually Sync LDAP Server ................................................................................................. 208
Figure 124: Ring Group Remote Extension .............................................................................................. 209
Figure 125: Paging/Intercom Group .......................................................................................................... 211
Figure 126: Page/Intercom Group Settings .............................................................................................. 212
Figure 127: Call Queue ............................................................................................................................. 213
Figure 128: Agent Login Settings .............................................................................................................. 216
Figure 129: Edit Extension Group ............................................................................................................. 217
Figure 130: Select Extension Group in Outbound Route .......................................................................... 218
Figure 131: Edit Pickup Group .................................................................................................................. 219
Figure 132: Edit Pickup Feature Code ...................................................................................................... 220
Figure 133: Music On Hold Default Class ................................................................................................. 221
Figure 134: Configure Analog Trunk without Fax Detection ..................................................................... 225
Figure 135: Configure Extension for Fax Machine: FXS Extension ......................................................... 225
Figure 136: Configure Extension for Fax Machine: Analog Settings ........................................................ 226
Figure 137: Configure Inbound Rule for Fax ............................................................................................. 226
Figure 138: Create Fax Extension ............................................................................................................ 227
Figure 139: Inbound Route to Fax Extension ........................................................................................... 228
Figure 140: Create Follow Me ................................................................................................................... 233
Figure 141: Edit Follow Me ....................................................................................................................... 233
Figure 142: Configure One-Key Dial ......................................................................................................... 237
Figure 143: One-Key Dial Destinations ..................................................................................................... 238
Figure 144: Create New DISA................................................................................................................... 239
Figure 145: Create New Event List ........................................................................................................... 244
Figure 146: Create Dial By Name Group .................................................................................................. 247
Figure 147: Dial By Name Group In IVR Key Pressing Events ................................................................ 248
Figure 148: Dial By Name Group In Inbound Rule ................................................................................... 249
Figure 149: Configure Extension First Name and Last Name .................................................................. 250
Figure 150: Status->PBX Status->Active Calls - Ringing .......................................................................... 251
Figure 151: Status->PBX Status->Active Calls – Call Established ........................................................... 251
Figure 152: Configure to Monitor an Active Call ....................................................................................... 252
Figure 153: Download Recording File from CDR Page ............................................................................ 259
Figure 154: Download Recording File from Recording Files Page ........................................................... 260
Figure 155: FXS Ports Signaling Preference ............................................................................................ 277
Figure 156: FXO Ports ACIM Settings ...................................................................................................... 277
Figure 157: Fax Sending in Web UI .......................................................................................................... 279
Figure 158: Announcements Center ......................................................................................................... 280
Figure 159: Announcements Center Group Configuration ........................................................................ 281
Figure 160: Announcements Center Code Configuration ......................................................................... 282
Figure 161: Announcements Center Example .......................................................................................... 282
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Figure 162: Status->PBX Status ............................................................................................................... 283
Figure 163: Trunk Status ........................................................................................................................... 283
Figure 164: Extension Status .................................................................................................................... 284
Figure 165: Queue Status ......................................................................................................................... 286
Figure 166: Conference Room Status ....................................................................................................... 287
Figure 167: UCM6204 Interfaces Status ................................................................................................... 287
Figure 168: Parking Lot Status .................................................................................................................. 288
Figure 169: System Status->Storage Usage ............................................................................................. 291
Figure 170: System Status->Resource Usage .......................................................................................... 291
Figure 171: System Events->Alert Events Lists: Disk Usage ................................................................... 292
Figure 172: System Events->Alert Events Lists: Memory Usage ............................................................. 293
Figure 173: System Events->Alert Events Lists: System Reboot ............................................................. 293
Figure 174: System Events->Alert Events Lists: System Crash ............................................................... 293
Figure 175: System Events->Alert Log ..................................................................................................... 294
Figure 176: Filter for Alert Log .................................................................................................................. 295
Figure 177: CDR Filter .............................................................................................................................. 296
Figure 178: Call Report ............................................................................................................................. 297
Figure 179: Call Report Entry with Audio Recording File .......................................................................... 297
Figure 180: Automatic Download Settings ................................................................................................ 298
Figure 181: CDR Report ........................................................................................................................... 298
Figure 182: Detailed CDR Information ...................................................................................................... 299
Figure 183: Downloaded CDR File Sample - Call To Shows "s" .............................................................. 299
Figure 184: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ................................ 300
Figure 185: Downloaded CDR File Sample - Source Channel and Dest Channel 2 ................................ 300
Figure 186: Downloaded CDR File Sample - Source Channel and Dest Channel 3 ................................ 300
Figure 187: CDR Statistics ........................................................................................................................ 301
Figure 188: CDR->Recording Files ........................................................................................................... 302
Figure 189: Network Upgrade ................................................................................................................... 305
Figure 190: Local Upgrade........................................................................................................................ 306
Figure 191: Upgrading Firmware Files ...................................................................................................... 307
Figure 192: Reboot UCM6200 .................................................................................................................. 307
Figure 193: Create New Backup ............................................................................................................... 309
Figure 194: Backup / Restore ................................................................................................................... 310
Figure 195: Local Backup ......................................................................................................................... 310
Figure 196: Data Sync .............................................................................................................................. 311
Figure 197: Restore UCM6200 from Backup File ..................................................................................... 312
Figure 198: Cleaner .................................................................................................................................. 313
Figure 199: Reset and Reboot .................................................................................................................. 314
Figure 200: Ethernet Capture.................................................................................................................... 316
Figure 201: PING ...................................................................................................................................... 316
Figure 202: Traceroute .............................................................................................................................. 317
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Figure 203: Troubleshooting Analog Trunks ............................................................................................. 318
Figure 204: Service Check ........................................................................................................................ 318
Figure 205: Network Status ....................................................................................................................... 319
Figure 206: SSH Access ........................................................................................................................... 320
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GNU GPL INFORMATION
UCM6200 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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CHANGE LOG
This section documents significant changes from previous versions of the UCM6200 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 1.0.0.1
This is the initial version.
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WELCOME
Thank you for purchasing Grandstream UCM6200 series IP PBX appliance. The UCM6200 series IP PBX
appliance is designed to bring enterprise-grade voice, video, data, and mobility features to
small-to-medium businesses (SMBs) in an easy-to-manage fashion. This IP PBX series allows businesses
to unify multiple communication technologies, such comprehensive voice, video calling, video
conferencing, video surveillance, data tools and facility access management onto one common network
that that can be managed and/or accessed remotely. The UCM6200 series supports a dual core 1GHz
ARM CortexTM A9 and 400Mhz VINETICTM A8 processors, 1GB RAM and 4GB flash. The secure and
reliable UCM6200 series delivers enterprise-grade features without any licensing fees, costs-per-feature
or recurring fees.
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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 UCM6200 as it may cause damage to the products
and void the manufacturer warranty.
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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
Table 1: Technical Specifications
Interfaces
Analog Telephone FXS Ports
2 ports (both with lifeline capability in case of power outage)
PSTN Line FXO Ports
UCM6202: 2 ports
UCM6204: 4 ports
UCM6208: 8 ports
Network Interfaces
UCM6202/6204: Dual Gigabit RJ45 ports with integrated PoE Plus
(IEEE 802.3at-2009)
UCM6208: Single Gigabit RJ45 port with integrated PoE Plus (IEEE
802.3at-2009)
NAT Router
Yes, UCM6204/UCM6208 only
Peripheral Ports
USB, SD
LED Indicators
Power/Ready, Network, PSTN Line, USB, SD
LCD Display
128x32 graphic LCD with DOWN and OK button
Reset Switch
Yes
Voice/Video Capabilities
Voice-over-Packet
Capabilities
LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier
grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection
and auto-switch to G.711
Voice and Fax Codecs
G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC
(30ms only), GSM, AAL2-G.726-32, ADPCM; T.38
Video Codecs
H.264, H.263, H.263+, VP8
QoS
Layer 3 QoS, Layer 2 QoS
Signaling and Control
DTMF Methods
In Audio, RFC2833, and SIP INFO
Provisioning Protocol and
Plug-and-Play
TFTP/HTTP/HTTPS, auto-discovery and 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, SIP (RFC3261), STUN, SRTP, TLS
Disconnect Methods
Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current
Disconnect, Busy Tone
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Security
Media
SRTP, TLS, HTTPS, SSH
Physical
Universal Power Supply
Output: 12VDC, 1.5A
Input: 100-240VAC, 50-60Hz
Dimensions
UCM6202/6204: 226mm (L) x 155mm (W) x 34.5mm (H)
UCM6208: 440mm (L) x 185mm (W) x 44mm (H)
Environmental
Operating: 32 - 104oF / 0 - 40oC, 10-90% (non-condensing)
Storage: 14 - 140oF / -10 - 60oC
Mounting
UCM6202/6204: Wall mount and Desktop
UCM6208: Rack mount and Desktop
Weight
UCM6202: Unit weight 0.51kg, Package weight 0.94kg
UCM6204: Unit weight 0.51kg, Package weight 0.94kg
UCM6208: Unit weight 2.23kg, Package weight 3.09kg
Additional Features
Multi-language Support
English/Simplified Chinese/Traditional Chinese/Spanish/French/
Portuguese/German/Russian/Italian/Polish/Czech for Web UI;
Customizable IVR/voice prompts for English, Chinese, British English,
German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese,
Russian, Swedish, Turkish, Hebrew, Arabic;
Customizable language pack to support any other languages
Caller ID
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 - BT
Polarity Reversal/ Wink
Yes, with enable/disable option upon call establishment and termination
Call Center
Multiple configurable call queues, automatic call distribution (ACD)
based on agent skills/availability busy level, in-queue announcement
Customizable Auto Attendant
Up to 5 layers of IVR (Interactive Voice Response)
Maximum Call Capacity
UCM6202: Concurrent audio calls up to 50, concurrent WebRTC
calls up to 25
UCM6204: Concurrent audio calls up to 75, concurrent WebRTC
calls up to 35
UCM6208: Concurrent audio calls up to 100, concurrent WebRTC
calls up to 50
Or up to 66% performance if calls are SRTP encrypted
Conference Bridges
UCM6202/6204: Up to 3 password-protected conference bridges
allowing up to 25 simultaneous PSTN or IP participants
UCM6208: Up to 6 password-protected conference bridges allowing
up to 32 simultaneous PSTN or IP participants
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Call Features
Call park, call forward, call transfer, DND, ring/hunt group,
paging/intercom and etc
Compliance
FCC: Part 15 (CFR 47) Class B, Part 68
CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3,
EN60950-1, TBR21, RoHS
A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS
60950, AS/ACIF S002 and ITU-T K.21 (Basic Level)
UL 60950 (power adapter)
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Note:
UCM6200 FXS ports lifeline functionality:
The UCM6200 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 UCM6200 series, the device needs to be properly powered up and
connected to network. This section describes detailed information on installation, connection and warranty
policy of the UCM6200 series.
EQUIPMENT PACKAGING
Table 2: UCM6202/UCM6204 Equipment Packaging
Main Case
Yes (1)
Power Adaptor
Yes (1)
Ethernet Cable
Yes (1)
Quick Installation Guide
Yes (1)
GPL License
Yes (1)
CONNECT YOUR UCM6200
CONNECT THE UCM6202
Figure 1: UCM6202 Front View
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Figure 2: UCM6202 Back View
To set up the UCM6202, follow the steps below:
1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6202.
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 UCM6202. Insert
the main plug of the power adapter into a surge-protected power outlet.
4. Wait for the UCM6202 to boot up. The LCD in the front will show the device hardware information
when the boot process is done.
5. Once the UCM6202 is successfully connected to network, the LED indicator for WAN in the front will
be in solid green and the LCD shows up 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.
CONNECT THE UCM6204
Figure 3: UCM6204 Front View
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Figure 4: UCM6204 Back View
To set up the UCM6204, follow the steps below:
1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6204.
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 UCM6204.
Insert the main plug of the power adapter into a surge-protected power outlet.
4. Wait for the UCM6204 to boot up. The LCD in the front will show the device hardware information
when the boot process is done.
5. Once the UCM6204 is successfully connected to network, the LED indicator for WAN in the front
will be in solid green and the LCD shows up 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.
CONNECT THE UCM6208
To set up the UCM6208, follow the steps below:
1. Connect one end of an RJ-45 Ethernet cable into the LAN port of the UCM6208.
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 UCM6208. Insert
the main plug of the power adapter into a surge-protected power outlet.
4. Wait for the UCM6208 to boot up. The LCD in the front will show the device hardware information
when the boot process is done.
5. Once the UCM6208 is successfully connected to network, the LED indicator for NETWORK in the front
will be in solid green and the LCD shows up the IP address.
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6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and
Fax) to the FXS ports.
Figure 5: UCM6208 Front View
Figure 6: UCM6208 Back View
SAFETY COMPLIANCES
The UCM6200 series IP PBX complies with FCC/CE and various safety standards. The UCM6200 power
adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM6200
package only. The manufacturer’s warranty does not cover damages to the device caused by unsupported
power adapters.
WARRANTY
If the UCM6200 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 a 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 UCM6200 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
The UCM6200 series provides LCD interface, LED indication and web GUI configuration interface.
The LCD displays hardware, software and network information. Users could also navigate in the LCD
menu for device information and basic network configuration.
The LED indication at the front of the device provides interface connection and activity status.
The web GUI gives users access to all the configurations and options for UCM6200 series setup.
This section provides step-by-step instructions on how to use the LCD menu, LED indicators and Web GUI
of the UCM6200 series. Once the basic settings are done, users could start making calls from UCM6200
extension registered on a SIP phone as described at the end of this section.
USE THE LCD MENU
Default LCD Display
When the device is powered up, the LCD will show device model (e.g., UCM6204), hardware version
(e.g., V1.0A) and IP address. Press "Down" button and the system time will be displayed as well.
Menu Access
Press "OK" button to start browsing menu options. Please see menu options in [Table 3: LCD Menu
Options].
Menu Navigation
Press the "Down" arrow key to browser different menu options. Press the "OK" button to select an
entry.
Exit
If "Back" option is available in the menu, select it to go back to the previous menu. For "Device Info"
"Network Info" and "Web Info" which do not have "Back" option, simply press the "OK" button to go
back to the previous menu. Also, the LCD will display default idle screen after staying in menu option
for 15 seconds.
LCD Backlight
The LCD backlight will be on upon key pressing. The backlight will go off after the LCD stays in idle for
30 seconds.
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Table 3: LCD Menu Options
View Events
Critical Events
Other Events
Device Info
Hardware: Hardware version number
Software: Software version number
P/N: Part number
WAN MAC: WAN side MAC address (UCM6202/UCM6204 only)
LAN MAC: LAN side MAC address
Uptime: System up time
Network Info
For UCM6208:
LAN Mode: DHCP, Static IP, or PPPoE
LAN IP: IP address
LAN Subnet Mask
For UCM6202/UCM6204:
WAN Mode: DHCP, Static IP, or PPPoE
WAN IP: IP address
WAN Subnet Mask
LAN IP: IP address
LAN Subnet Mask
Network Menu
For UCM6208:
LAN Mode: Select LAN mode as DHCP, Static IP or PPPoE
Static Route Reset: Click to reset the static route setting
For UCM6202/UCM6204:
WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE
Static Route Reset: Click to reset the static route setting
Factory Menu
Reboot
Factory Reset
LCD Test Patterns
Press "OK" to start. Then press "Down" button to test different LCD
patterns. When done, press "OK" button to exit.
Fan Mode
Select "Auto" or "On".
LED Test Patterns
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Select "All On" "All Off" or "Blinking" and check LED status.
RTC Test Patterns
Select "2022-02-22 22:22" or "2011-01-11 11:11" to start the RTC
(Real-Time Clock) test pattern. Then check the system time from LCD idle
screen by pressing "DOWN" button, or from web GUI->System
Status->General page. Reboot the device manually after the RTC test is
done.
Hardware Testing
Select "Test SVIP" to perform SVIP test on the device. This is mainly for
factory testing purpose which verifies the hardware connection inside the
device. The diagnostic result will display in the LCD after the test is done.
Web Info
Protocol: Web access protocol. HTTP or HTTPS. By default it's HTTPS
Port: Web access port number. By default it's 8089
SSH Switch
Enable SSH: Enable SSH access.
Disable SSH: Disable SSH access.
By default the SSH access is disabled.
USE THE LED INDICATORS
The UCM6200 has LED indicators in the front to display connection status. The following table shows the
status definitions.
Table 4: UCM6202/UCM6204 LED INDICATORS
LED Indicator
LED Status
LAN
WAN
USB
SD
FXS (Phone/Fax)
FXO (Telco Line)
Solid: Connected
Flashing: Data Transferring
OFF: Not Connected
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Table 5: UCM6208 LED INDICATORS
LED
LED Status
NETWORK
Solid: Connected
OFF: Not Connected
ACT
USB
SD
Phone (FXS)
Line (FXO)
Solid: Connected
Flashing: Data Transferring
OFF: Not Connected
USE THE WEB GUI
ACCESS WEB GUI
The UCM6200 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow users to configure the device through a Web browser such as Microsoft IE, Mozilla Firefox,
Google Chrome and etc.
Figure 7: UCM6204 Web GUI Login Page
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To access the Web GUI:
1. Connect the computer to the same network as the UCM6200.
2. Ensure the device is properly powered up and shows its IP address on the LCD.
3. Open a Web browser on the computer and enter the web GUI URL in the following format:
http(s)://IP-Address:Port
where the IP-Address is the IP address displayed on the UCM6200 LCD.
By default, the protocol is HTTPS and the Port number is 8089.
For example, if the LCD shows 192.168.40.167, please enter the following in your web browser:
https://192.168.40.167:8089
4. Enter the administrator’s login and password to access the Web Configuration Menu. The default
administrator's username and password is "admin" and "admin". It is highly recommended to change
the default password after login for the first time.
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Note:
By default, the UCM6200 has "Redirect From Port 80" enabled. Therefore, if users type in the UCM6200 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, please enter 192.168.40.167 in your web
browser and the web page will be redirected to:
https://192.168.40.167:8089
The option "Redirect From Port 80" can be configured under the UCM6200 web GUI->Settings->HTTP
Server.
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SETUP WIZARD
When the user logs in the UCM6200 web UI for the first time, a setup wizard will guide the user to set up basic
configuration. Configurations in setup wizard includes: Time zone, Change password, Network settings,
Extensions, Trunk and routes.
Figure 8: UCM6200 Setup Wizard
During the wizard, the user can quit the setup wizard at any time to start over with manual configuration. At
the last step of the wizard, the user will be provided with summary for review, before the configuration is
loaded. Once the setup is completed, the system is ready to go.
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WEB GUI CONFIGURATIONS
There are four main sections in the Web GUI for users to view the PBX status, configure and manage the
PBX.
Status: Displays PBX status, System Status, System Events and CDR.
PBX: To configure extensions, trunks, call routes, zero config for auto provisioning, call features,
internal options, IAX settings and SIP settings.
Settings: To configure user management, network settings, firewall settings, change password,
LDAP Server, HTTP Server, Email Settings, Time Settings, NTP server, recording storage and login
timeout.
Maintenance: To perform firmware upgrade, backup configurations, cleaner setup, reset/reboot,
syslog setup and troubleshooting.
WEB GUI LANGUAGES
Currently the UCM6200 series web GUI supports English, Simplified Chinese, Traditional Chinese,
Spanish, French, Portuguese, Russian, Italian, Polish, German and etc.
Users can select the displayed language in web GUI login page, or at the upper right of the web GUI after
logging in.
Figure 9: UCM6200 Web GUI Language
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SAVE AND APPLY CHANGES
Click on "Save" button after configuring the web GUI options in one page. After saving all the changes,
make sure click on "Apply Changes" button on the upper right of the web page to submit all the changes. If
the change requires reboot to take effect, a prompted message will pop up for you to reboot the device.
MAKE YOUR FIRST CALL
Power up the UCM6200 and your SIP end point phone. Connect both devices to the network. Then follow
the steps below to make your first call.
1. Log in the UCM6200 web GUI, go to PBX->Basic/Call Routes->Extensions.
2. Click on "Create New SIP Extension" to create a new extension. You will need User ID, Password and
Voicemail Password information to register and use the extension later.
3. Register the extension on your phone with the SIP User ID, SIP server and SIP Password information.
The SIP server address is the UCM6200 IP address.
4. When your phone is registered with the extension, dial *97 to access the voicemail box. Enter the
Voicemail Password once you hear "Password" voice prompt.
5. Once successfully logged in to the voicemail, you will be prompted with the Voice Mail Main menu.
6. You are successfully connected to the PBX system now.
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SYSTEM SETTINGS
This section explains configurations for system-wide parameters on the UCM6200. System settings are
under “Settings” tag on UCM6200 web GUI. System settings include User Management, Network Settings,
Firewall, Change Password, LDAP server, HTTP server, Email settings, Time Settings, NTP Server,
Recordings Storage and Login Timeout settings.
USER MANAGEMENT
User management is on web GUI->Settings->User Management page. User could create multiple
accounts for different administrators to log in the UCM6200 web GUI. Additionally, the system will
automatically create user accounts along with creating new extensions for extension users to login to the
web UI using their extension number and password. All existing user accounts for web UI login will be
displayed on User Management page as shown in the following figure.
Figure 10: User Management Page Display
USER PRIVILEGES
Three privilege levels are supported:
Super Admin
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- This is the highest privilege. Super Admin can access all pages on UCM6200 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’s the only
user with “Super Admin” privilege. The User Name and Privilege level cannot be changed or
deleted.
- Super Admin could change its own login password on web UI->Settings->Change Password
page.
- Super Admin could view operations done by all the users in web UI->Settings->User
Management->Operation Log.
Admin
- Users with “Admin” privilege can only be created by “Super Admin” user.
- “Admin” privilege users are not allowed to access the following pages:
Maintenance->Upgrade
Maintenance->Backup
Maintenance->Cleaner
Maintenance->Reset/Reboot
Settings->User Management->Operation Log
- “Admin” privilege users cannot create new users for login.
Consumer
- A user account for web UI login is created automatically by the system when a new extension is
created.
- The user could log in the web UI with the extension number and password to access user
information, extension configuration and CDR of that extension.
CREATE NEW WEB UI USER
When logged in as Super Admin, click on to create a new account for web UI user. The
following dialog will prompt. Configure the parameters as shown in below table.
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Figure 11: Create New User
Table 6: User Management->Create New User
User Name
Configure a username to identify the user which will be required in web UI login.
Letters, digits and underscore are allowed in the user name.
User Password
Configure a password for this user which will be required in web UI login. Letters,
digits and underscore are allowed.
Privilege
This is the role of the web UI user. Currently only “Admin” is supported when
Super Admin creates a new user.
Department
Enter the necessary information to keep a record for this user.
Fax
Email Address
First Name
Last Name
Home Number
Phone Number
Once created, the Super Admin can edit the users by clicking on or delete the user by clicking on .
Figure 12: User Management – New Users
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USER PORTAL
The user could log in web UI user portal using the extension number and password. When there is an
extension created in the UCM6200, the corresponding user account for the extension is automatically
created. The user portal allows limited access including user information, extension configuration and CDR
information of the extension. The login username is the extension number and the password is configured
by Super Admin. The following figure shows the dialog of editing the account information by Super Admin.
The User Name must be the extension number and it’s not configurable.
Figure 13: Edit User Information by Super Admin
The following figure shows an example of login page using extension number 1000 as the username.
Figure 14: User Portal Login
After login, the web UI displays is shown as below.
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Figure 15: User Portal Layout
For the configuration parameter information in each page, please refer to [Table 6: User
Management->Create New User] for options in User Portal->Basic Information->User Information page;
please refer to [EXTENSIONS] for options in User Portal->Basic Information->Extension page; please
refer to [CDR] for User Portal->Basic Information->CDR page.
CONCURRENT MULTI-USER LOGIN
When there are multiple web UI users created, concurrent multi-user login is supported on the UCM6200.
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.
Figure 16: Multiple User Operation Error Prompt
OPERATION LOG
Super Admin has the authority to view operation logs on UCM6200 web GUI->Settings->User
Management->Operation Log page. Operation logs list operations done by all the web UI users, for
example, web UI login, creating trunk, creating outbound rule and etc. There are 6 columns to record the
operation details “Date”, “User Name”, “IP Address”, “Results”, “Page Operation” and “Specific Operation”.
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Figure 17: Operation Logs
The operation log can be sorted and filtered for easy access. Click on the header of each column to sort.
For example, clicking on "Date" will sort the logs according to operation date and time. Clicking on "Date"
again will reverse the order.
Table 7: Operation Log Column Header
Date
The date and time when the operation is executed.
User Name
The username of the user who performed the operation.
IP Address
The IP address from which the operation is made.
Results
The result of the operation.
Page 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.
User could also filter the operation logs by time condition, IP address and/or username. Configure these
conditions and then click on .
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Figure 18: 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.
CHANGE PASSWORD
After logging in the UCM6200 web UI for the first time, it is highly recommended for users to change the
default password "admin" to a more complicated password for security purpose. Follow the steps below to
change the Web UI access password.
1. Go to Web UI->Settings->User Management-> Change Password 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.
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Figure 19 : Change Password
Enter Old Password
Enter the Old Password for UCM6200
Enter New Password
Enter the New Password for UCM6200
Retype New
Password
Retype the New Password for UCM6200
Email Address
Configure the Email address for UCM6200. In case login credential is lost, Email
address is used to retrieve login credential
CHANGE BINDNG EMAIL
UCM6200 allows user to configure binding email in case login password is lost. UCM6200 login credential
will be sent to the designated email address. The feature can be found under web UI->Settings->User
Management->Change Binding Email.
Figure 20: Change Binding Email
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Table 8: Change Binding Email option
Enter the password
of the account
Enter the current login user credential for UCM6200
Email Address
Email Address is used to retrieve password when password is lost
NETWORK SETTINGS
After successfully connecting the UCM6200 to the network for the first time, users could login the Web GUI
and go to Settings->Network Settings to configure the network parameters for the device.
Additional network functions and settings are available for UCM6202 and UCM6204:
UCM6202/UCM6204 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
GUI->Settings->Network Settings page to configure LAN settings, WAN settings (UCM6202/UCM6204
only), 802.1X and Port Forwarding (UCM6202/UCM6204 only).
BASIC SETTINGS
Please refer to the following tables for basic network configuration parameters on UCM6202/UCM6204,
and UCM6208 respectively.
Table 9: UCM6202/UCM6204 Network Settings->Basic Settings
Method
Select "Route", "Switch" or "Dual" mode on the network interface of UCM6200.
The default setting is "Route".
Route
WAN port interface will be used for uplink connection. LAN port interface will
be used to serve as router.
Switch
WAN port interface will be used for uplink connection. LAN port interface will
be used as bridge for PC connection.
Dual
Both ports can be 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" for this interface.
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Preferred DNS Server
Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to
use it as Primary DNS 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. The default setting is
0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
User Name
Enter the user name 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
Enable or disable DHCP server capability. The default setting is "Yes".
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 Address
From
Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.
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.
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. The default setting is
0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
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User Name
Enter the user name 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 UCM6202 WAN port) or LAN 2 (mapped to UCM6202 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. The default setting is
0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
User Name
Enter the user name 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.
Table 10: UCM6208 Network Settings->Basic Settings
Preferred DNS Server
Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to
use it as Primary DNS server.
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings.
Subnet Mask
Enter the subnet mask address for static IP settings.
Gateway IP
Enter the gateway IP address for static IP settings.
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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.
User Name
Enter the user name 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.
Method: Route
When the UCM6200 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 21: UCM6200 Network Interface Method: Route
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Method: Switch
WAN port interface is used for uplink connection; LAN port interface is used as bridge for PC
connection.
Figure 22: UCM6200 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 23: UCM6200 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 is allowed to access Internet or other LAN resources. The
UCM6200 supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure
show UCM6200 uses 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.
Figure 24: UCM6200 Using 802.1X as Client
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Figure 25: UCM6200 Using 802.1X EAP-MD5
The following table shows the configuration parameters for 802.1X on UCM6200. 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 as the 802.1X mode, users
will also need upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated
from the RADIUS server.
Table 11: UCM6200 Network Settings->802.1X
802.1X Mode
Select 802.1X mode. The default setting is "Disable". The supported 802.1X
mode are:
EAP-MD5
EAP-TLS
EAP-PEAPv0/MSCHAPv2
Identity
Enter 802.1X mode identity information.
MD5 Password
Enter 802.1X mode MD5 password information.
802.1X Certificate
Select 802.1X certificate from local PC and then upload.
802.1X Client
Certificate
Select 802.1X client certificate from local PC and then upload.
STATIC ROUTES
The UCM6200 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 UCM6200 web
GUI->Network Settings->Basic 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 UCM6200 as a
failover backup, and etc.
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Click on to create a new 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.
Table 12: UCM6200 Network Settings->Static Routes
Destination
Configure the destination IP address or the destination IP subnet for the
UCM6200 to reach using the static route.
Example:
IP address - 192.168.66.4
IP subnet - 192.168.66.0
Netmask
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
Gateway
Configure the gateway address so that the UCM6200 can reach the destination
via this gateway. Gateway address is optional.
Example:
192.168.40.5
Interface
Specify the network interface on the UCM6200 to reach the destination using the
static route.
LAN interface is eth0; WAN interface is eth1.
Static routes configuration can be reset from LCD menu->Network Menu.
The following diagram shows a sample application of static route usage on UCM6204.
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Figure 26: UCM6204 Static Route Sample
The network topology of the above diagram is as below:
Network 192.168.69.0 has IP phones registered to UCM6204 LAN 1 address
Network 192.168.40.0 has IP phones registered to UCM6204 LAN 2 address
Network 192.168.66.0 has IP phones registered to UCM6204 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 UCM6204. Therefore, we need configure a
static route on the UCM6204 so that the phones in isolated networks can make calls between each other.
Figure 27: UCM6204 Static Route Configuration
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PORT FORWORDING
The UCM network interface supports router function which provides users the ability to do port forwarding.
If the UCM6202/UCM6204 LAN mode is set to "Route" under web GUI->Settings->Network
Settings->Basic Settings page, port forwarding is available for configuration.
The port forwarding configuration is under web GUI->Settings->Network Settings->Port Forwarding
page. Please see related settings in the table below.
Table 13: UCM6202/UCM6204 Network Settings->Port Forwarding
WAN Port
Specify the WAN port number or a range of WAN ports. Up to 8 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.
LAN IP
Specify the LAN IP address.
LAN Port
Specify the LAN port number or a range of LAN ports.
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.
Protocol Type
Select protocol type "UDP Only", "TCP Only" or "TCP/UDP" for the forwarding in
the selected port. The default setting is "UDP Only".
The following figures demonstrate a port forwarding example to provide phone’s web UI access to public
side.
The UCM6202/UCM6204 network mode is set to “Route”
The UCM6202/UCM6204 WAN port is connected to uplink switch, with a public IP address configured,
e.g. 1.1.1.1.
The UCM6202/UCM6204 LAN port provides DHCP pool that connects to multiple phone devices in the
LAN network 192.168.2.x. The UCM6202/UCM6204 is used as a router, with gateway address
192.168.2.1
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There is a GXP2160 connected under the LAN interface network of the UCM6202/UCM6204. It
obtains IP address 192.168.2.100 from UCM6200 DHCP pool
On the UCM6202/UCM6204 web UI->Settings->Network Settings->Port Forwarding, configure a
port forwarding entry as the figure shows below.
WAN Port: This is the port opened up on the WAN side for access purpose.
LAN IP: This is the GXP2160 IP address, under the LAN interface network of the
UCM6202/UCM6204.
Protocol Type: We select TCP here for web UI access using HTTP.
Figure 28: UCM6202/UCM6204 Port Forwarding Configuration
This will allow users to access the GXP2160 web UI from public side, by typing in address “1.1.1.1:8088”.
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Figure 29: GXP2160 Web Access Using UCM6202 Port Forwarding
DDNS SETTINGS
DDNS setting allows user to access UCM6200 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 UCM6200 needs to have public IP access.
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Figure 30: Register Domain Name on noip.com
2. On web UI->Settings->Network Settings->DDNS Settings, enable DDNS service and configure
username, password and host name.
Figure 31: UCM6200 DDNS Setting
3. Now you can use domain name instead of IP address to connect to the UCM6200 web UI.
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Figure 32: Using Domain Name to Connect to UCM6200
FIREWALL
The UCM6200 provides users firewall configurations to prevent certain malicious attack to the UCM6200
system. Users could configure to allow, restrict or reject specific traffic through the device for security and
bandwidth purpose. The UCM6200 also provides Fail2ban feature for authentication errors in SIP
REGISTER, INVITE and SUBSCRIBE. To configure firewall settings in the UCM6200, go to Web
UI->Settings->Firewall page.
STATIC DEFENSE
Under Web GUI->Settings->Firewall->Static 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 UCM6200.
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Table 14: UCM6200 Firewall->Static Defense->Current Service
Port
Process
Type
Protocol or Service
7777
Asterisk
tcp/IPv4
SIP
389
Slapd
tcp/IPv4
LDAP
22
Dropbear
tcp/IPv4
SSH
80
Lighthttpd
tcp/IPv4
HTTP
8089
Lighthttpd
tcp/IPv4
HTTPS
69
Opentftpd
udp/IPv4
TFTP
9090
Asterisk
udp/IPv4
SIP
6060
zero_config
udp/IPv4
UCM6200 zero_config service
5060
Asterisk
udp/IPv4
SIP
4569
Asterisk
udp/IPv4
SIP
5353
zero_config
udp/IPv4
UCM6200 zero_config service
37435
Syslogd
udp/IPv4
Syslog
For typical firewall settings, users could configure the following options on the UCM6200.
Table 15: Typical Firewall Settings
Ping Defense
Enable
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 (UCM6202/UCM6204) interface.
Ping-of-Death
Defense Enable
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
(UCM6202/UCM6204) interface.
Under "Custom Firewall Settings", users could create new rules to accept, reject or drop certain traffic
going through the UCM6200. 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’s 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
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Figure 33: Create New Firewall Rule
Table 16: Firewall Rule Settings
Rule Name
Specify the Firewall rule name to identify the firewall rule.
Action
Select the action for the Firewall to perform.
ACCEPT
REJECT
DROP
Type
Select the traffic type.
IN
If selected, users will need specify the network interface "LAN" or "WAN"
(for UCM6202/UCM6204) for the incoming traffic.
OUT
Service
Select the service type.
FTP
SSH
Telnet
TFTP
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".
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:
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Click on to edit the rule
Click on to delete the rule
DYNAMIC DEFENSE
Dynamic defense is supported on the UCM6200 series. It can blacklist hosts dynamically when the LAN
mode is set to "Route" under web GUI->Settings->Network Settings->Basic Settings page. If enabled,
the traffic coming into the UCM6200 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 UCM6200 firewall,
which will then be displayed in the web page. Please refer to the following table for dynamic defense
options on the UCM6200.
Table 17: UCM6200 Firewall Dynamic Defense
Dynamic Defense
Enable
Enable dynamic defense. The default setting is disabled.
Periodical Time
Interval
Configure the dynamic defense periodic time interval (in minutes). If the
number of TCP connections from a host exceeds the connection threshold
within this period, this host will be added into Blacklist. The valid value is
between 1 and 59 when dynamic defense is turned on. The default setting is
59.
Blacklist Update
Interval
Configure the blacklist update time interval (in seconds). The default setting is
120.
Connection
Threshold
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.
Dynamic Defense
Whitelist
Configure the dynamic defense whitelist.
For example,
192.168.1.3
192.168.1.4
The following figure shows a configuration example like this:
If a host at IP address 192.168.40.7 initiates more than 20 TCP connections to the UCM6200 within 1
minute, it will be added into UCM6200 blacklist.
This host 192.168.40.7 will be blocked by the UCM6200 for 300 seconds.
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Since IP address 192.168.40.5 is in whitelist, if the host at IP address 192.168.40.5 initiates more than
20 TCP connections to the UCM6200 within 1 minute, it will not be added into UCM6200 blacklist. It
can still establish TCP connection with the UCM6200.
Figure 34: Configure Dynamic Defense
FAIL2BAN
Fail2Ban feature on the UCM6200 provides intrusion detection and prevention for authentication errors in
SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within "Max Retry Duration", the
UCM6200 will take action to forbid the host for certain period as defined in "Banned Duration". This feature
helps prevent SIP brute force attacks to the PBX system.
Table 18: Fail2Ban Settings
Global Settings
Enable Fail2Ban
Enable Fail2Ban. The default setting is disabled. Please make sure both "Enable
Fail2Ban" and "Asterisk Service" are turned on in order to use Fail2Ban for SIP
authentication on the UCM6200.
Banned Duration
Configure the duration (in seconds) for the detected host to be banned. The
default setting is 300. If set to -1, the host will be always banned.
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 5.
MaxRetry
Configure the number of authentication failures during "Max Retry Duration"
before the host is banned. The default setting is 10.
Fail2Ban Whitelist
Configure IP address, CIDR mask or DNS host in the whiltelist. Fail2Ban will not
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ban the host with matching address in this list. Up to 5 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 in order to use
Fail2Ban for SIP authentication on the UCM6200.
Protocol
Configure the listening port number for the service. Currently only 5060 (for UDP)
is supported.
MaxRetry
Configure the number of authentication failures during "Max Retry Duration"
before the host is banned. The default setting is 10. Please make sure this option
is properly configured as it will override the "MaxRetry" value under "Global
Settings".
LDAP SERVER
The UCM6200 has an embedded LDAP 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 UCM6200 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 UCM6200 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.
If users have the Grandstream phone provisioned by the UCM6200, 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 UCM6200. If the UCM6200 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
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UCM6200 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 other LDAP server.
To access LDAP server and client settings, go to Web GUI->Settings->LDAP Server.
LDAP SERVER CONFIGURATIONS
The following figure shows the default LDAP server configurations on the UCM6200.
Figure 35: LDAP Server Configurations
The UCM6200 LDAP server supports anonymous access (read-only) by default. Therefore the LDAP client
doesn't 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 for the first
phonebook under LDAP Phonebook.
Figure 36: Default LDAP Phonebook DN
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Figure 37: Default LDAP Phonebook Attributes
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 GUI->PBX->Basic/Call Routes->Extensions first. The default LDAP phonebook will then be
updated automatically.
Figure 38: LDAP Server->LDAP Phonebook
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Add new phonebook
A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under
"LDAP Phonebook" section.
Figure 39: 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.
Figure 40: Edit LDAP Phonebook
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Import phonebook from your computer to LDAP server
Click on “Import Phonebook” and a dialog will prompt as shown in the figure below.
Figure 41: Import Phonebook
The file to be imported must be a CSV file with UTF-8 encoding. Users can open the CSV 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 UCM6200 LDAP phonebook section first and
use it as a sample to start with.
Figure 42: Phonebook CSV File Format
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 UCM6200 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 UCM6200 LDAP Phonebook, the contacts in the CSV file will be added into
the existing phonebook. If the phonebook DN doesn’t exist on the UCM6200 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
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UCM6200.
Figure 43: 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 UCM6200 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
file for the users to add more contacts in it and import to the UCM6200 again.
Figure 44: Export Selected LDAP Phonebook
LDAP CLIENT CONFIGURATIONS
The configuration on LDAP client is similar when you use other LDAP servers. Here we provide an
example on how to configure the LDAP client on the SIP end points to use the default PBX phonebook.
Assuming the server base dn is "dc=pbx,dc=com", configure the LDAP clients as follows (case
insensitive):
Server Address: LDAP server IP address
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Base DN: dc=pbx,dc=com
User Name: cn= “LDAP server login name”, dc=pbx, dc=com [matching LDAP server format]
Password: “LDAP server login password”
Filter: (|(CallerIDName=%)(AccountNumber=%))
Port: 389
The following figure gives a sample configurations for UCM6200 acting as a LDAP client.
Figure 45: LDAP Client Configurations
To configure Grandstream IP phones as the LDAP client, please refer to the following example:
Server Address: The IP address or domain name of the UCM6200
Base DN: dc=pbx,dc=com
User Name: 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 GXP2200 to successfully use
the LDAP server as configured in Figure 35: LDAP Server Configurations.
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Figure 46: GXP2200 LDAP Phonebook Configuration
HTTP SERVER
The UCM6200 embedded web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow the users to configure the PBX through a Web browser such as Microsoft IE, Mozilla Firefox
and Google Chrome. By default, the PBX can be accessed via HTTPS using Port 8089 (e.g.,
https://192.168.40.50:8089). Users could also change the access protocol and port as preferred under
Web GUI->Settings->HTTP Server.
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Table 19: HTTP Server Settings
Redirect From Port 80
Enable or disable redirect from port 80. On the PBX, the default access
protocol is HTTPS and the default port number is 8089. When this option
is enabled, the access using HTTP with Port 80 will be redirected to
HTTPS with Port 8089. The default setting is "Enable".
Protocol Type
Select HTTP or HTTPS. The default setting is "HTTPS". This is also the
protocol used for zero config when the end point device downloads the
config file from the UCM6200.
Port
Specify port number to access the HTTP server. The default port number
is 8089.
Once the change is saved, the web page will be redirected to the login page using the new URL. Enter the
username and password to login again.
EMAIL SETTINGS
The Email application on the UCM6200 can be used to send out alert event Emails, Fax (Fax-To-Email),
Voicemail (Voicemail-To-Email) and etc. The configuration parameters can be accessed via Web
GUI->Settings->Email Settings.
Table 20: Email Settings
TLS Enable
Enable or disable TLS during transferring/submitting your Email to other
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".
Server
Specify the SMTP server when using type "Client".
Username
Username is required when using type "Client". Normally it's the Email
address.
Password
Password to login for the above Username (Email address) is required
when using type "Client".
Display Name
Specify the display name in the FROM header in the Email.
Sender
Specify the sender's Email address.
For example, pbx@example.mycompany.com.
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The following figure shows a sample Email settings on the UCM6200, assuming the Email is using
smtp.gmail.com as the SMTP server.
Figure 47: UCM6200 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 UCM6200.
TIME SETTINGS
AUTO TIME UPDATING
The current system time on the UCM6200 is displayed on the upper right of the web page. It can also be
found under Web GUI->Status->System Status->General.
To configure the UCM6200 to update time automatically, go to Web GUI->Settings->Time
Settings->Time Auto Updating.
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Note:
The configurations under Web GUI->Settings->Time Settings->Time Auto Updating page require
reboot to take effect. Please consider configuring auto time updating related changes when setting up the
UCM6200 for the first time to avoid service interrupt after installation and deployment in production.
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Table 21: Time Auto Updating
Remote NTP Server
Specify the URL or IP address of the NTP server for the UCM6200 to
synchronize the date and time. The default NTP server is
ntp.ipvideotalk.com.
Enable DHCP Option 2
If set to "Yes", the UCM6200 is allowed to get provisioned for Time Zone
from DHCP Option 2 in the local server automatically. The default setting
is "Yes".
Enable DHCP Option 42
If set to "Yes", the UCM6200 is allowed to 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".
Time Zone
Select the proper time zone option so the UCM6200 can display correct
time accordingly.
If "Self-Defined Tome Zone" is selected, please specify the time zone
parameters in "Self-Defined Time Zone" field as described in below
option.
Self-Defined Time Zone
If "Self-Defined Time Zone" is selected in "Time Zone" option, users will
need define their own time zone following the format below.
The syntax is: std offset dst [offset], start [/time], end [/time]
Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0
MTZ+6MDT+5
This indicates a time zone with 6 hours offset and 1 hour ahead for DST,
which is U.S central time. If it is positive (+), the local time zone is west of
the Prime Meridian (A.K.A: International or Greenwich Meridian); If it is
negative (-), the local time zone is east.
M4.1.0,M11.1.0
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec).
The 2nd number indicates the nth iteration of the weekday: (1st Sunday,
3rd Tuesday…). Normally 1, 2, 3, 4 are used. If 5 is used, it means the
last iteration of the weekday.
The 3rd number indicates weekday: 0,1,2,..,6 ( for Sun, Mon,
Tues, ... ,Sat).
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
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SET TIME MANUALLY
To manually set the time on the UCM6200, go to Web UI->Settings->Time Settings->Set Time Manually.
The format is YYYY-MM-DD HH:MI:SS.
Figure 48: Set Time Manually
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Note:
Manually setup time will take effect immediately after saving and applying change in the web UI. If users
would like to reboot the UCM6200 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
GUI->Settings->Time Settings->Time Auto 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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OFFICE TIME
On the UCM6200, 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 UI->Settings->Time Settings->Office Time. Click on "Create New Office Time" to create an office
time.
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Figure 49: Create New Office Time
Table 22: Create New Office Time
Start Time
Configure the start time for office hour.
End Time
Configure the end time for office hour
Week
Select the work days in one week.
Show Advanced Options
Check this options to show advanced options. Once selected, please
specify "Month" and "Day" below.
Month
Select the months for office time.
Day
Select the work days in one month.
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.
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Figure 50: Settings->Time Settings->Office Time
Click on to edit the office time.
Click on to delete the office time.
Click on "Delete Selected Office Times" to delete multiple selected office times at once.
HOLIDAY
On the UCM6200, 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 UI->Settings->Time Settings->Holiday. Click on "Create New Holiday" to create holiday time.
Figure 51: Create New Holiday
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Table 23: Create New Holiday
Name
Specify the holiday name to identify this holiday.
Holiday Memo
Create a note for the holiday.
Month
Select the month for the holiday.
Day
Select the day for the holiday.
Show Advanced Options
Check this option to show advanced options. If selected, please specify
the days as holiday in one week below.
Week
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.
Figure 52: Settings->Time Settings->Holiday
Click on to edit the holiday.
Click on to delete the holiday.
Click on "Delete Selected Holidays" to delete multiple selected holidays at once.
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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/How_to_use_office_time_and_holiday_UCM62
00.pdf
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NTP SERVER
The UCM6200 can be used as a NTP server for the NTP clients to synchronize their time with. To
configure the UCM6200 as the NTP server, set "Enable NTP server" to "Yes" under web
GUI->Settings->NTP Server. On the client side, point the NTP server address to the UCM6200 IP
address or host name to use the UCM6200 as the NTP server.
RECORDINGS STORAGE
The UCM6200 supports call recordings automatically or manually and the recording files can be saved in
external storage plugged in the UCM6200 or on the UCM6200 locally. To manage the recording storage,
users can go to UCM6200 web GUI->Settings->Recordings Storage page and select whether to store
the recording files in USB Disk, SD card or locally on the UCM6200.
Figure 53: Settings->Recordings Storage
If “Enable Auto Change” is selected, the recording files will be automatically saved in the available
USB Disk or SD card plugged into the UCM6200. 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 UCM6200 internal storage.
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 UCM6200.
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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 54: 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.
Figure 55: Recording Storage Category
On the UCM6200, 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.
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LOGIN TIMEOUT SETTINGS
After the user logs in the UCM6200 web UI, the user will be automatically logged out after certain timeout.
This timeout value can be specified under UCM100 web GUI->Settings->Login Timeout Settings page.
The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user doesn’t make
any operation on web UI within the timeout, the user will be logged out automatically. After that, the web UI
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 UI login session and the user will not be automatically logged out.
Figure 56: Login Timeout Settings
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GOOGLE SERVICE SETTINGS SUPPORT
UCM6200 now supports Google OAuth 2.0 authentication. This feature is used for supporting UCM6200
conference scheduling system. Once OAuth 2.0 is enabled, UCM6200 conference system can access Google
calendar to schedule or update conference.
Google Service Settings can be found under web GUI-> Settings-> Google Service Settings-> Google
Service Settings.
Figure 57: Google Service Settings->OAuth2.0 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 UCM6200
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.
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Figure 58: Google Service->New Project
2. Enable Calendar API from API Library.
3. Click “Credentials” on the left drop down menu to create new OAuth2.0 login credentials.
Figure 59: Google Service->Create 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.
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Figure 60: Google Service->OAuth2.0 Login
6. Now UCM6200 is connected with Google Service.
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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 UCM6200 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 UCM6200. The settings for this feature can be accessed via Web
UI->PBX->Zero 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 61: 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 UCM6200 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 UCM6200 for auto provisioning. Three methods of
auto provisioning are used.
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Figure 62: UCM6200 Zero Config
SIP SUBSCRIBE
When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The
UCM6200 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 UCM6200 and take the
new configuration.
DHCP OPTION 66
This method should be used on the UCM6202/UCM6204 because only the UCM6202/UCM6204 have
WAN and LAN port with LAN port supporting the router function. When the phone restarts (by default
DHCP Option 66 is turned on), it will send out a DHCP DISCOVER request. The UCM6202 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 UCM6200.
mDNS
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When the phone boots up, it sends out mDNS query to get the TFTP server address. The UCM6200
will respond with its own address. The phone will then send TFTP request to download the XML config
file from the UCM6200.
To start the auto provisioning process, under Web GUI->PBX->Zero Config->Zero Config Settings, fill in
the auto provision information.
Figure 63: Auto Provision Settings
Table 24: Auto Provision Settings
Enable Zero Config
Enable or disable the zero config feature on the PBX. The default setting
is enabled.
Enable Automatic
Configuration Assignment
By default, this is disabled. If disabled, when SIP device boots up, the
UCM6200 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
UCM6200.
Note:
When disabled, SIP devices can still be provisioned by manually sending
NOTIFY from the UCM6200 which will include the XML config file URL for
the SIP device to download.
Automatically Assign
Extension
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.
Zero Config Extension
Click on the link "Zero Config Extension Segment" to specify the
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Segment
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 UI->PBX->Internal
Options->General page->Extension Preference section: "Auto Provision
Extensions".
Enable Pick Extension
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.
Pick Extension Segment
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 UI->PBX->Internal
Options->General page->Extension Preference section: "Pick
Extensions".
Pick Extension Period
(hour):
Specify the number of minutes to allow the phones being provisioned to
pick extensions.
Please make sure an extension is manually assigned to the phone or "Automatically Assign Extension" is
enabled during provisioning. After the configuration on the UCM6200 web GUI, click on "Save" and "Apply
Changes". Once the phone boots up and picks up the config file from the UCM6200, it will take the
configuration right away.
DISCOVERY
Users could manually discover the device by specifying the IP address or scanning the entire LAN network.
Three methods are supported to scan the devices.
PING
ARP
SIP Message (NOTIFY)
Click on "Auto Discover" under web UI-> PBX-> Zero Config->Zero 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 UCM6200. 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 UCM6200 web
GUI->PBX->Zero Config->Auto Provisioning Settings.
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Figure 64: 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) are
displayed in the list.
Figure 65: Discovered Devices
GLOBAL CONFIGURATION
GLOBAL POLICY
Global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN
with the UCM6200 no matter what the Grandstream device model it is. It is divided into two levels:
Web UI->PBX->Zero Config->Global Policy
Web UI->PBX->Zero Config->Global Templates.
Global Templates configuration has higher priority to Global Policy configuration.
Global Policy can be accessed in web GUI->PBX->Zero Config->Global 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:
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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.
Select the checkbox on the left of the parameter you would like to configure to active the dropdown list for
this parameter.
Figure 66: Global Policy Categories
The following tables list the Global Policy configuration parameters for the SIP end device.
Table 25: Global Policy Parameters->Localization
Language settings
Language
Select the LCD display language on the SIP end device.
Date and Time
Date Format
Configure the date display format on the SIP end device’s LCD.
Time Format
Configure the time display in 12-hour or 24-hour format on the SIP end
device’s LCD.
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.
Time Zone
Configure the time zone used on the SIP end device.
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Table 26: 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 which NAT traversal mechanism will be enabled on the
endpoint device.
If set to "STUN" and STUN server is configured, the phone system will
periodically send STUN message to the SUTN server to get the public IP
address of its NAT environment and keep the NAT port open. STUN will
not work if the NAT is symmetric type.
If set to “Keep-alive”, the phone system will send the STUN packets to
maintain the connection that is first established during registration of the
phone. The “Keep-alive” packets will fool the NAT device into keeping the
connection open and this allows the host server to send SIP requests
directly to the registered phone.
If it needs to use OpenVPN to connect host server, it needs to set it to
“VPN”.
If the firewall and the SIP device behind the firewall are both able to use
UPNP, it can be set to “UPNP”. The both parties will negotiate to use
which port to allow SIP through.
The default setting is "Keep-alive".
Use Random Port
Configure whether to allow the endpoint device to use random ports for
both SIP and RTP messages. This is usually necessary when multiple
phones are behind the same full cone NAT. The default setting is "No".
Note: This parameter must be set to "No" for Direct IP Calling to work.
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]]];
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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.
Table 27: Global Policy Parameters->Contact List
LDAP Phonebook
Source
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 UCM6200
web UI->Settings->LDAP Server will be applied.
Address
Configure the IP address or DNS name of the LDAP server.
Port
Configure the LDAP server port. The default value is 389.
Base DN
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
User Name
Configure the bind "Username" for querying LDAP servers. The field can
be left blank if the LDAP server allows anonymous binds.
Password
Configure the bind "Password" for querying LDAP servers. The field can
be left blank if the LDAP server allows anonymous binds.
Number Filter
Configure the filter used for number lookups. Please refer to user manual
for more details.
Name Filter
Configure the filter used for name lookups. Please refer to user manual
for more details.
Version
Select the protocol version for the phone to send the bind requests. The
default value is 3.
Name Attribute
Specify the "name" attributes of each record which are returned in the
LDAP search result. Example:
gn
cn sn description
Number Attribute
Specify the "number" attributes of each record which are returned in the
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LDAP search result. Example:
telephoneNumber
telephoneNumber Mobile
Display Name
Configure the entry information to be shown on phone's LCD. Up to 3
fields can be displayed. Example:
%cn %sn %telephoneNumber
Max Hits
Specify the maximum number of results to be returned by the LDAP
server. Valid range is 1 to 3000. The default value is 50.
Search Timeout
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. The default value
is 30.
Sort Results
Specify whether the searching result is sorted or not. The default setting is
No.
Incoming Calls
Configure to enable LDAP number searching when receiving calls. The
default setting is No.
Outgoing Calls
Configure to enable LDAP number searching when making calls. The
default setting is No.
Lookup Display Name
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.
XML Phonebook
Phonebook XML Server
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.
Phonebook Download
Interval
Configure the phonebook download interval (in Minute). If set to 0,
automatic download will be disabled. Valid range is 5 to 720.
Remove manually-edited
entries on download
If set to "Yes", when XML phonebook is downloaded, the entries added
manually will be automatically removed.
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Table 28: Global Policy Parameters->Maintenance
Upgrade and Provision
Firmware Source
Firmware source via ZeroConfig provisioning could a URL for external
server address, local UCM directory or USB media if plugged in to the
UCM6200.
Select a source to get the firmware file:
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 UCM6200 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
UCM6200 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 UCM6200 and
the firmware file must be put under a folder named “ZC_firmware” in
the USB storage root directory.
Upgrade via
When URL is selected as firmware source, configure upgrade via TFTP,
HTTP or HTTPS.
Server Path
When URL is selected as firmware source, configure the firmware
upgrading server path.
File Prefix
When URL is selected as firmware source, configure the firmware file
prefix. If configured, only the firmware with the matching encrypted prefix
will be downloaded and flashed into the phone, if URL is selected as
firmware source.
File Postfix
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.
Allow DHCP Option 43/66
If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can
be redirected.
Automatic Upgrade
If enabled, the endpoint device will automatically upgrade if a new
firmware is detected. Users can select automatic upgrading by day, by
week or by minute.
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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.
By minute
Once selected, specify the interval X that the SIP end device will
request for new firmware every X minutes.
Firmware Upgrade Rule
Specify how firmware upgrading and provisioning request to be sent.
Web Access
Admin Password
Configure the administrator password for admin level login.
End-User Password
Configure the end-user password for the end user level login.
Web Access Mode
Select HTTP or HTTPS as the web access protocol.
Web Server Port
Configure the port for web access. The valid range is 1 to 65535.
Security
Disable Telnet/SSH
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.
Syslog
Syslog Server
Configure the URL/IP address for the syslog server.
Syslog Level
Select the level of logging for syslog.
Send SIP Log
Configure whether the SIP log will be included in the syslog message.
Table 29: Global Policy Parameters->Network Settings
Basic Settings
IP Address
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 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.
Advanced Setting
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Layer 3 QoS
Define the Layer 3 QoS parameter. This value is used for IP Precedence,
Diff-Serv or MPLS. Valid range is 0-63.
Layer 2 QoS Tag
Assign the VLAN Tag of the Layer 2 QoS packets. Valid range is 0 -4095.
Layer 2 QoS Priority Value
Assign the priority value of the Layer 2 QoS packets. Valid range is 0-7.
STUN Server
Configure the IP address or Domain name of the STUN server. Only
non-symmetric NAT routers work with STUN.
Keep Alive Interval
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.
Table 30: Global Policy Parameters->Customization
Wallpaper
Screen Resolution 1024 x
600
Check this option if the SIP end device shall use 1024 x 600 resolution for
the LCD screen wallpaper.
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 UCM6200.
Screen Resolution 800 x 400
Check this option if the SIP end device shall use 800 x 400 resolution for
the LCD screen wallpaper.
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 UCM6200.
Screen Resolution 480 x 272
Check this option if the SIP end device shall use 480 x 272 resolution for
the LCD screen wallpaper.
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 UCM6200.
Screen Resolution 320 x 240
Check this option if the SIP end device supports 320 x 240 resolution for
the LCD screen wallpaper.
Source
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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 UCM6200.
GLOBAL TEMPLATES
Global Templates can be accessed in web GUI->PBX->Zero Config->Global 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 dialog->Advanced Settings, the user can select to use
one of the global template 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 "Create New Template" to add a global template. Users will see the following configurations.
Table 31: Create New Template
Template Name
Create a name to identify this global template.
Description
Provide a description for the global template. This is optional.
Active
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 and click on “Add Option” to add it into the global template.
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Figure 67: 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 remove 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 UI->PBX->Zero Config->Global Templates page.
Users can click on to delete the global template or click on “Delete Selected Templates” to delete
multiple selected templates at once.
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Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected
templates.
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 PBX->Zero Config->Model
Templates. If multiple model templates are created and enabled, when the user configures the device in
Edit Device dialog->Advanced Settings, the user can select to use one of the model template 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 UCM6200.
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 “Create New Template” to add a model template.
Table 32: Create New Model Template
Model
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
Enter a description for the model template. This is optional.
Default Model Template
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.
Active
Check this option to enable the model template.
Click on to edit the model template.
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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 and click on “Add Option” to add it into the model template.
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 68: Edit Model Template
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Click on Save when done. The model template will be displayed on web UI->PBX->Zero
Config->Model 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.
MODEL UPDATE
UCM6200 zero config feature supports provisioning all models of Grandstream SIP end devices.
Templates for most of the Grandstream models are built in with the UCM6200 already. Templates for GS
Wave and Grandstream surveillance products require users to download and install under web
UI->PBX->Zero Config->Model Update first before they are available in the UCM6200 for selection. After
downloading and installing the model template to the UCM6200, it will show in the dropdown list for
“Model” selection when editing the model template.
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 UCM6200.
Figure 69: Template Management
In case the UCM6200 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 UI. Please contact Grandstream customer support if
the model package is needed for manual uploading.
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Figure 70: Upload Model Template Manually
DEVICE CONFIGURATION
On web GUI, page PBX->Zero Config->Zero Config, users could create new device, delete existing
device(s), make special configuration for a single device, or send NOTIFY to existing device(s).
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 UCM6200. 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 "Create New Device" 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 “Create New Device” to save the configuration for this device.
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Figure 71: Create New Device
MANAGE DEVICES
The device manually created or discovered from Auto Discover will be listed in the web UI->PBX->Zero
Config->Zero Config page. Users can see the devices with their MAC address, IP address, vendor,
model and etc.
Figure 72: Manage Devices
Click on to access the web UI of the phone.
Click on to edit the device configuration.
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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 73: 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.
(1) Global Policy
This is the lowest level configuration. The global policy configured in web UI->PBX->Zero
Config->Global Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page
PBX->Zero Config->Global Policy.
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(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 UI->PBX->Zero Config->Model Templates page.
Please see default model template option in [Table 32: Create New Model Template].
(4) Model Templates
Select a model 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 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 74: 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 following 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.
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Figure 75: Add P Value in Customize Device Settings
Select multiple devices that need to be modified and then click on 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 76: 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.
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Figure 77: Modify Selected Devices - Different Models
--------------------------------------------------------------------------------------------------------------------------------------------
Note:
Performing batch operation will override all the existing device configuration on the page.
--------------------------------------------------------------------------------------------------------------------------------------------
After the above configurations, save the changes and go back to web UI->PBX->Zero Config->Zero
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.
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Figure 78: 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 GUI->PBX->Zero Config->Zero Config Settings, select “Enable Zero Config”.
2. Go to web GUI->PBX->Zero Config->Global Policy, configure Date Format, Time Format and
Firmware Source as follows.
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Figure 79: Zero Config Sample - Global Policy
3. Go to web GUI->PBX->Zero Config->Model Templates, create a new model template “English
Support Template” for GXP2140. 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 GUI->PBX->Zero Config->Model Templates, create another model template “Spanish
Support Template” for GXP2140. 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 UI->PBX->Zero
Config->Zero Config.
6. On web GUI->PBX->Zero Config->Zero 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.
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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 80: 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.
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Figure 81: Zero Config Sample - Device Preview 2
Select “Spanish Support Template” in ○
4 “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.
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Figure 82: Zero Config Sample - Device Preview 3
10. Click on “Apply Changes” to apply saved changes.
11. On the web UI->PBX->Zero Config->Zero Config page, click on to send NOTIFY to trigger the
device to download config file from UCM6200.
Now all the 9 phones in the network will be provisioned with an unique extension registered on the
UCM6200. 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’s 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 GUI->PBX->Basic/Call Routes->Extensions. Click on
"Create New User"->"Create New SIP Extension" and a new dialog window will show for users to fill in the
extension information.
Figure 83: Create New Device
SIP extension options are divided into four categories:
Basic Settings
Media
Features
Specific Time
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Click on the tag to view or edit options belonging to that category.
The configuration parameters are as follows.
Table 33: SIP Extension Configuration Parameters->Basic 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.
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 using this rule.
SIP/IAX Password
Configure the password for the user. A random secure password will be
automatically generated. It is recommended to use this password for
security purpose.
Auth ID
Configure the authentication ID for the user. If not configured, the
extension number will be used for authentication.
Enable Voicemail
Enable voicemail for the user. The default setting is "Yes".
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.
Disable This Extension
If selected, this extension will be disabled on the UCM6200.
Note:
The disabled extension still exists on the PBX but can’t be used on the
end device.
User Settings
First Name
Configure the first name of the user. The first name can contain
characters, letters, digits and _.
Last Name
Configure the last name of the user. The last name can contain
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characters, letters, digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
User Password
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.
Language
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 GUI->PBX->Internal Options->Language. The dropdown list
shows all the current available voice prompt languages on the UCM6200.
To add more languages in the list, please download voice prompt
package by selecting "Check Prompt List" under web UI->PBX->Internal
Options->Language.
Concurrent Registrations
The maximum endpoints which can be registered into this extension. For
security concerns, the default value is 1.
Table 34: SIP Extension Configuration Parameters->Media
SIP Settings
NAT
Use NAT when the UCM6200 is on a public IP communicating with
devices hidden behind NAT (e.g., broadband router). If there is one-way
audio issue, usually it's related to NAT configuration or Firewall's support
of SIP and RTP ports. The default setting is enabled.
Can Direct Media
By default, the UCM6200 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 UCM6200 to negotiate endpoint-to-endpoint media routing. The
default setting is "No".
DTMF Mode
Select DTMF mode for the user to send DTMF. The default setting is
"RFC2833". If "Info" is selected, SIP INFO message will be used. If
"Inband" is selected, 64-kbit PCMU and PCMA are required. When "Auto"
is selected, RFC2833 will be used if offered, otherwise "Inband" will be
used.
TEL URI
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.
Enable Keep-alive
If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is "Yes".
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Keep-alive Frequency
Configure the Keep-alive interval (in seconds) to check if the host is up.
The default setting is 60 seconds.
Enable T.38 UDPTL
Enable or disable T.38 UDPTL support.
SRTP
Enable SRTP for the call. The default setting is disabled.
Fax Mode
Select Fax mode. The default setting is “None”.
None: Disable Fax.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Strategy
This option controls how the extension can be used on devices within
different types of network.
Allow All
Device in any network can register this extension.
Local Subnet Only
Only the user in specific subnet can register this extension. Up to
three subnet addresses can be specified.
A Specific IP Address
Only the device on the specific IP address can register this extension.
The default setting is "Allow All".
Codec Preference
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.263, H.263p and VP8.
Table 35: SIP Extension Configuration Parameters->Features
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 condition 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
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time.
Office Time and Holiday could be configured on page Settings->Time
Settings->Office 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
condition 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 Settings->Time
Settings->Office Time/Holiday page.
Call Forward Busy
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 condition
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 Settings->Time
Settings->Office Time/Holiday page.
CC Settings
Enable CC
If enabled, UCM6200 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’s disabled.
CC Mode
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
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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.
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.
Time Condition for Ring
Simultaneously
Ring the external number simultaneously along with the extension on the
basis of this time condition.
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
UCM6200, which can be configured in the global ring timeout setting
under web GUI->Internal Options->IVR 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
GUI->CDR->Recording 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 Skip
Trunk Auth
If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which
users can skip entering password when making outbound calls.
Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Support Hot-Desking Mode
If enabled, SIP Password will accept only alphabet characters and digits.
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Auth ID will be changed to the same as Extension.
Enable LDAP
If enabled, the extension will be added to LDAP Phonebook PBX list.
Enable WebRTC Support
Enable registration and call from WebRTC.
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.
Table 36: SIP Extension Configuration Parameters->Specific Time
Specific Time
Time Condition
Click to add Time Condition to configure specific time for this extension.
CREATE NEW IAX EXTENSION
The UCM6200 supports Inter-Asterisk eXchange (IAX) protocol. IAX is used for transporting VoIP
telephony sessions between servers and terminal devices. IAX is similar to SIP but also has its own
characteristic. For more information, please refer to RFC 5465.
To manually create new IAX user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on
"Create New User"->"Create New IAX Extension" and a new dialog window will show for users to fill in the
extension information. The configuration parameters are as follows.
Table 37: IAX Extension Configuration Parameters->Basic 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.
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 using this rule.
SIP/IAX Password
Configure the password for the user. A random secure password will be
automatically generated. It is recommended to use this password for
security purpose.
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Enable Voicemail
Enable voicemail for the user. The default setting is "Yes".
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.
Disable This Extension
If selected, this extension will be disabled on the UCM6200.
Note:
The disabled extension still exists on the PBX but can’t be used on the
end device.
User Settings
First Name
Configure the first name of the user. The first name can contain
characters, letters, digits and _.
Last Name
Configure the last name of the user. The last name can contain
characters, letters, digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
User Password
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.
Language
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 GUI->PBX->Internal Options->Language. The dropdown list
shows all the current available voice prompt languages on the UCM6200.
To add more languages in the list, please download voice prompt
package by selecting "Check Prompt List" under web UI->PBX->Internal
Options->Language.
Table 38: IAX Extension Configuration Parameters->Media
SIP 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. The
default setting is "Yes".
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SRTP
Enable SRTP for the call. The default setting is disabled.
Fax Mode
Select Fax Mode. The default setting is “None”.
None: Disable Fax. This is the default setting.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Strategy
This option controls how the extension can be used on devices within
different types of network.
Allow All
Device in any network can register this extension.
Local Subnet Only
Only the user in specific subnet can register this extension. Up to
three subnet addresses can be specified.
A Specific IP Address
Only the device on the specific IP address can register this extension.
The default setting is "Allow All".
Codec Preference
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.263, H.263p and VP8.
Table 39: IAX Extension Configuration Parameters->Features
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 condition 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 Settings->Time
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Settings->Office 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
condition 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 Settings->Time
Settings->Office Time/Holiday page.
Call Forward Busy
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 condition
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 Settings->Time
Settings->Office Time/Holiday page.
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.
Other Settings
Ring Timeout
Configure the number of seconds to ring the user before the call is
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forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6200, which can be configured in the global ring timeout setting
under web GUI->Internal Options->IVR 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
GUI->CDR->Recording 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 Skip
Trunk Auth
If “Skip Trunk Auth” is set to “By Time”, select a time condition during
which users can 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.
Call Duration Limit
The maximum duration of call-blocking.
Table 40: IAX Extension Configuration Parameters->Specific Time
Specific Time
Time Condition
Click to add Time Condition to configure specific time for this extension.
CREATE NEW FXS EXTENSION
The UCM6200 supports Foreign eXchange Subscriber (FXS) interface. FXS is used when user needs to
connect analog phone lines or FAX machines to the UCM6200.
To manually create new FXS user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on
"Create New User"->"Create New FXS Extension" and a new dialog window will show for users to fill in the
extension information. The configuration parameters are as follows.
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Table 41: FXS Extension Configuration Parameters->Basic Settings
General
Extension
The extension number associated with the user.
Analog Station
Select the FXS port to be assigned for this extension.
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.
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 using this rule.
Enable Voicemail
Enable voicemail for the user. The default setting is "Yes".
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.
Disable This Extension
If selected, this extension will be disabled on the UCM6200.
Note:
The disabled extension still exists on the PBX but can’t be used on the
end device.
User Settings
First Name
Configure the first name of the user. The first name can contain
characters, letters, digits and _.
Last Name
Configure the last name of the user. The last name can contain
characters, letters, digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
User Password
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.
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Language
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 GUI->PBX->Internal Options->Language. The dropdown list
shows all the current available voice prompt languages on the UCM6200.
To add more languages in the list, please download voice prompt
package by selecting "Check Prompt List" under web UI->PBX->Internal
Options->Language.
Table 42: FXS Extension Configuration Parameters->Media
Analog Settings
Call Waiting
Configure to enable/disable call waiting feature. The default setting is
"No".
User ‘#’ as SEND
If configured, the # key can be used as SNED key after dialing the
number on the analog phone. The default setting is "Yes".
RX Gain
Configure the RX gain for the receiving channel of analog FXS port. The
valid range is -30dB to +6dB. The default setting is 0.
TX Gain
Configure the TX gain for the transmitting channel of analog FXS port.
The valid range is -30dB to +6dB. The default setting is 0.
MIN RX Flash
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.
MAX RX Flash
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 can't be
modified. The default setting is 1250ms.
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 "Yes".
Echo Cancellation
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 means 256/8 = 32 ms.
The default setting is "ON", which is 128 taps.
3-Way Calling
Configure to enable/disable 3-way calling feature on the user. The default
setting is enabled.
Send CallerID After
Configure the number of rings before sending CID. Default setting is 1.
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Fax Mode
For FXS extension, there are three options available in Fax Mode. The
default setting is “None”.
None: Disable Fax.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Fax Gateway: If selected, the UCM6200 can support conversation
and processing of Fax data from T.30 to T.38 or T.38 to T.30. This
feature is only available for FXS or FXO port.
Table 43: FXS Extension Configuration Parameters->Features
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 condition 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 Settings->Time
Settings->Office 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
condition 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
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time.
Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
Call Forward Busy
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 condition
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 Settings->Time
Settings->Office Time/Holiday page.
CC Settings
Enable CC
If enabled, UCM6200 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.
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
Hotline Type
Configure the Hotline Type:
Immediate Hotline: When the phone is off-hook, UCM6200 will
immediately dial the preset number
Delay Hotline: When the phone is off-hook, if there is no dialing
within 5 seconds, UCM6200 will dial the preset number.
Other Settings
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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
UCM6200, which can be configured in the global ring timeout setting
under web GUI->Internal Options->IVR 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
GUI->CDR->Recording 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 Skip
Trunk Auth
If “Skip Trunk Auth” is set to “By Time”, select a time condition during
which users can 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.
Music On Hold
Select which Music On Hold class to suggest to extension when putting
the active call on hold.
Call Duration Limit
Configure the maximum duration of call-blocking.
Table 44: FXS Extension Configuration Parameters->Specific Time
Specific Time
Time Condition
Click to add Time Condition to configure specific time for this extension.
BATCH ADD EXTENSIONS
BATCH ADD SIP EXTENSIONS
In order to add multiple SIP extensions, BATCH add can be used to create standardized SIP extension
accounts. However, unique extension user name can’t be set using BATCH add.
Under Web GUI->PBX->Basic/Call Routes->Extensions, click on "Batch Add Extensions"->"Batch Add
SIP Extensions".
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Table 45: Batch Add SIP Extension Parameters
General
Start Extension
Configure the starting extension number of the batch of extensions to be
added.
Create Number
Specify the number of extensions to be added. The default setting is 5.
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.
Enable Voicemail
Enable Voicemail for the user. The default setting is "Yes".
SIP/IAX Password
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.
Voicemail Password
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.
Enter a password to be used on all the extensions in the batch.
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
UCM6200, which can be configured in the global ring timeout setting
under web GUI->Internal Options->IVR 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
GUI->CDR->Recording 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
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option is disabled.
Music On Hold
Select which Music On Hold class to suggest to extensions when putting
them on hold.
Enable LDAP
If enabled, the batch added extensions will be added to LDAP Phonebook
PBX list; if disabled, the batch added extensions will be skipped when
creating LDAP Phonebook.
Enable WebRTC Support
If enabled, extensions will be able to login to user portal and use Web
RTC features.
Call Duration Limit
Configure the maximum duration of call-blocking.
SIP Settings
NAT
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's related to NAT configuration or Firewall's support of SIP
and RTP ports. The default setting is enabled.
Can 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".
DTMF Mode
Select DTMF mode for the user to send DTMF. The default setting is
"RFC2833". If "Info" is selected, SIP INFO message will be used. If
"Inband" is selected, 64-kbit codec PCMU and PCMA are required. When
"Auto" is selected, RFC2833 will be used if offered, otherwise "Inband"
will be used.
Enable Keep-alive
If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is "Yes".
Keep-alive Frequency
Configure the number of seconds for the host to be up for Keep-alive. The
default setting is 60 seconds.
TEL URI
If the end device/phone 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-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.
Concurrent Registrations
The maximum endpoints which can be registered into this extension. For
security concerns, the default value is 1.
Other Settings
SRTP
Enable SRTP for the call. The default setting is "No".
Fax Mode
Select Fax mode for this user. The default setting is “None”.
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None: Disable Fax.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Strategy
This option controls how the extension can be used on devices within
different types of network.
Allow All
Device in any network can register this extension.
Local Subnet Only
Only the user in specific subnet can register this extension. Up to
three subnet addresses can be specified.
A Specific IP Address.
Only the device on the specific IP address can register this extension.
The default setting is "Allow All".
Enable T.38 UDPTL
Enable or disable T.38 UDPTL Support.
Skip Trunk Auth
If enable “All”, users do not need to enter password when making an
outbound call. If enable “Follow Me”, the user can dial out via follow me
without password.
Codec Preference
Select audio and video codec for the extension. The available codecs are:
PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, Ilbc, ADPCM,
LPC10, H.264, H.263, H.263p and VP8.
BATCH ADD IAX EXTENSIONS
Under Web GUI->PBX->Basic/Call Routes->Extensions, click on “Batch Add Extensions”->”Batch Add
IAX Extensions”.
Table 46: Batch Add IAX Extension Parameters
General
Start Extension
Configure the starting extension number of the batch of extensions to be
added.
Create Number
Specify the number of extensions to be added. The default setting is 5.
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:
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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.
Enable Voicemail
Enable Voicemail for the user. The default setting is “Yes”.
SIP/IAX Password
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.
Voicemail Password
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.
Enter a password to be used on all the extensions in the batch.
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
UCM6200, which can be configured in the global ring timeout setting
under web GUI->Internal Options->IVR 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
GUI->CDR->Recording 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.
Music On Hold
Select which Music On Hold class to suggest to extensions when putting
them on hold.
Enable LDAP
If enabled, the batch added extensions will be added to LDAP Phonebook
PBX list; if disabled, the batch added extensions will be skipped when
creating LDAP Phonebook.
Call Duration Limit
Configure the maximum duration of call-blocking.
IAX Settings
Max Number of Calls
Configure the maximum number of calls allowed for each remote IP
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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. The
default setting is "Yes".
Other Settings
SRTP
Enable SRTP for the call. The default setting is "No".
Fax Mode
Select Fax Mode for this user. The default setting is “None”.
None: Disable Fax.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Strategy
This option controls how the extension can be used on devices within
different types of network.
Allow All
Device in any network can register this extension.
Local Subnet Only
Only the user in specific subnet can register this extension. Up to
three subnet addresses can be specified.
A Specific IP Address.
Only the device on the specific IP address can register this extension.
The default setting is "Allow All".
Skip Trunk Auth
If enable “All”, users do not need to enter password when making an
outbound call. If enable “Follow Me”, the call can dial out via follow me
without password.
Codec Preference
Select audio and video codec for the extension. The available codecs are:
PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, iLBC,
ADPCM, LPC10, H.264, H.263, H.263p and VP8.
SEARCH AND EDIT EXTENSION
All the UCM6200 extensions are listed under Web GUI->PBX->Basic/Call Routes->Extensions, with
status, Extension, CallerID Name, Technology (SIP, IAX and FXS), IP and Port. Each extension has a
checkbox for users to "Modify Selected Extensions" or "Delete Selected Extensions". 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.
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Figure 84: Manage Extensions
Status
Users can see the following icon for each extension to indicate the SIP status.
Green: Free
Blue: Ringing
Yellow: In Use
Grey: Unavailable (the extension is not registered or disabled on the PBX)
Edit single extension
Click on to start editing the extension parameters.
Reboot the user
Click on to send NOTIFY reboot event to the device which has an UCM6200 extension already
registered. To successfully reboot the user, "Zero Config" needs to be enabled on the UCM6200 web
GUI->PBX->Zero Config->Auto Provisioning 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 "Modify Selected Extensions" to edit the
extensions in a batch.
Delete selected extensions
Select the checkbox for the extension(s). Then click on "Delete Selected Extensions" to delete the
extension(s).
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EXPORT EXTENSIONS
The extensions configured on the UCM6200 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.
Figure 85: Export Extensions
The exported csv file can be serve as a template for users to fill in desired extension information to be
imported to the UCM6200.
IMPORT EXTENSIONS
The capability to import extensions to the UCM6200 provides users flexibility to batch add extensions with
similar or different configuration quickly into the PBX system.
1. Export extension csv file from the UCM6200 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 86: 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
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information as previously configured without change.
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 to select csv file from local directory in the PC.
6. Click on "Save" to import the csv file.
7. Click on "Apply Changes" to apply the imported file on the UCM6200.
EMAIL TO USER
Once the extensions are created with Email address, the PBX administrator can click on button “Email To
User” to send the account registration and configuration information to the user. Please make sure Email
setting under web UI->Settings->Email Settings is properly configured and tested on the UCM6200
before using “Email To User”.
When click on “Email To User” 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 87: Email To User - Prompt Information
The user will receive Email including account registration information and LDAP configuration. A QR code
is also generated for Mobile applications to scan it and get automatically provisioned. QR code
provisioning is supported on Grandstream Softphone GS Wave AndroidTM application and iOS application.
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Figure 88: Account Registration Information and QR Code
Figure 89: LDAP Client Information and QR Code
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MULTIPLE REGISTRATIONS PER EXTENSION
UCM6200 supports multiple registrations per extension so that users can use the same extension on
devices in different locations.
Figure 90: Multiple Registrations per Extension
This feature can be enabled by configuring option “Concurrent Registrations” under web
UI->PBX->Basic/Call Routes->Edit Extension. The default value is set to 1 for security purpose.
Figure 91: Extension - Concurrent Registration
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SMS MESSAGE SUPPORT
The UCM6200 provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or
GXV phones that supports SIP message, after an UCM6200 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.
SMS Message support is a new feature added since firmware 1.0.10.x.
Figure 92: SMS Message Support
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TRUNKS
ANALOG TRUNKS
Go to Web GUI->PBX->Basic/Call Routes->Analog 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 47: Analog Trunk Configuration Parameters
Channels
Select the channel for the analog trunk.
UCM6202: 2 channels
UCM6204: 4 channels
UCM6208: 8 channels
Trunk Name
Specify a unique label to identify the trunk when listed in outbound rules,
incoming rules and etc.
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 or not other stations are allowed to
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.
Advanced Options
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
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default setting is “No”.
Polarity on Answer Delay
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.
Current Disconnect
Threshold (ms)
This is the periodic time (in ms) that the UCM6200 will use to check on a
voltage drop in the line. The default setting is 200. The valid range is 50 to
3000.
Ring Timeout
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
UCM6200 considers a non-ringing line with hangup activity. The default
setting is 8000.
RX Gain
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.
TX Gain
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.
Use CallerID
Configure to enable CallerID detection. The default setting is “Yes”.
Caller ID Scheme
Select the Caller ID scheme for this trunk. The default setting is
“Bellcore/Telcordia”.
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
GUI->CDR->Recording Files.
Disable This Trunk
If selected, the trunk will be disabled.
DAHDI Out Line Selection
This is to implement analog trunk outbound line selection strategy. Three
options are available:
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
would be port 1->port 2->port 10->port 16. Every time it will start with
port 1 (if it's 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
1->2->10->16->1->2->10->16->1->2->10->16..., following the last
port being used.
Descend
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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
would be port 16->port 10->port 2->port 1. Every time it will start with
port 16 (if it's idle).
The default setting is “Ascend” mode.
Tone Settings
Busy Detection
Busy Detection is used to detect far end hangup or for detecting busy
signal. The default setting is "Yes".
Busy Tone Count
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
Congestion detection is used to detect far end congestion signal. The
default setting is "Yes".
Congestion Count
If "Congestion Detection" is enabled, users can specify the number of
congestion tones to wait for. The default setting is 2.
Tone Country
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)".
Busy Tone
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
Congestion Tone
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=250/250
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PSTN Detection
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 UCM6200 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
UCM6200 settings.
1. Go to UCM6200 web GUI->PBX->Basic/Call Routes->Analog Trunks page.
2. Click to edit the analog trunk created for the FXO port.
3. In the dialog 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 93: UCM6200 FXO Tone Settings
4. Click on "Detect" to start PSTN detection.
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Figure 94: UCM6200 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.
Figure 95: UCM6200 PSTN Detection: Auto Detect
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If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.
Figure 96: UCM6200 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.
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 UCM6200 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
UCM6200 settings.
Table 48: PSTN Detection for Analog Trunk
Detect Model
Select "Auto Detect" or "Semi-auto Detect" for PSTN detection.
Auto Detect
Please make sure two or more channels are connected to the
UCM6200 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
UCM6200 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
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manually. Please make sure one channel is connected to the
UCM6200 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.
The default setting is "Auto Detect".
Source Channel
Select the channel to be detected.
Destination Channel
Select the channel to help detect when "Auto Detect" is used.
Destination Number
Configure the number to be called to help the detection.
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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 trunks can be configured in UCM6200 under Web GUI->PBX->Basic/Call Routes->VoIP 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 "Create New SIP Trunk" or "Create New IAX Trunk" to add a new VoIP trunk.
Click on to configure detailed parameters for the VoIP trunk.
Click on to configure Direct Outward Dialing (DOD) for the SIP Trunk.
Click on to start LDAP Sync.
Click on to delete the VoIP trunk.
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For VoIP trunk example, please refer to the document in the following link:
http://www.grandstream.com/sites/default/files/Resources/ucm_to_ucm_peer_guide.pdf
The VoIP trunk options are listed in the table below.
Table 49: Create New SIP Trunk
Type
Select the VoIP trunk type.
Peer SIP Trunk
Register SIP Trunk
Provider Name
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the
trunk.
Keep Original CID
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.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension’s CID when
the extension has CID configured. The default setting is “No”.
NAT
Turn on this setting 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 support on the firewall.
Disable This Trunk
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.
TEL URI
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-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.
Caller ID
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.
When making outgoing calls, the following rules are used to determine
which CallerID will be used if they exist:
The CallerID configured for the extension will be looked up first.
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If no CallerID is configured for the extension, the CallerID configured
for the trunk will be used.
If the above two are missing, the "Global Outbound CID" defined in
Web GUI->PBX->Internal Options->General will be used.
Need Registration
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.
Username
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 from the provider when
"Register SIP Trunk" is selected.
Auth ID
Enter the Authentication ID for "Register SIP Trunk" type.
Auto Record
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 GUI->CDR->Recording Files.
Table 50: SIP Register Trunk Configuration Parameters
Basic Settings
Provider Name
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the
trunk.
Transport
Configure the SIP transport protocol to be used in this trunk. The default
setting is "All - UDP Primary".
UDP Only
TCP Only
TLS Only
All - UDP Primary: UDP is the primary transport protocol when all the
other SIP transport methods are available too.
All - TCP Primary: TCP is the primary transport protocol when all the
other SIP transport methods are available too.
All – TLS Primary: TLS is the primary transport protocol when all the
other SIP transport methods are available too.
Keep Original CID
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.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension’s CID when
the extension has CID configured. The default setting is “No”.
NAT
Turn on this option when the PBX is using public IP and communicating
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with devices behind NAT. If there is one-way audio issue, usually it’s
related to NAT configuration or SIP/RTP port configuration on the firewall.
Disable This Trunk
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.
TEL URI
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-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.
Need Registration
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.
Username
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 from the provider when
"Register SIP Trunk" is selected.
Auth ID
Enter the Authentication ID for "Register SIP Trunk" type.
Auto Record
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 GUI->CDR->Recording Files.
Advanced Settings
Codec Preference
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.263, H.263p and VP8.
From Domain
Configure the actual domain name where the extension comes from. This
can be used to override the From Header.
For example, "trunk.UCM6200.provider.com" is the From Domain in From
Header: sip:1234567@trunk.UCM6200.provider.com.
From User
Configure the actual user name 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.UCM6200.provider.com.
Send PPI Header
If enabled, the SIP INVITE message sent to the trunk will contain PPI
(P-Preferred-Identity) header. The default setting is “No”.
Note:
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“Send PPI Header” and “Send PAI Header” cannot be enabled at the
same time. Only one of the two headers is allowed to be contained in the
SIP INVITE message.
Send PAI Header
If enabled, the SIP INVITE message sent to the trunk will contain PAI
(P-Asserted-Identity) 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 is allowed to be contained in the
SIP INVITE message.
Outbound Proxy Support
Select to enable outbound proxy in this trunk. The default setting is "No".
Outbound Proxy
When outbound proxy support is enabled, enter the IP address or URL of
the outbound proxy.
DID Mode
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".
DTMF Mode
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 UI->PBX->SIP
Settings->ToS.
RFC2833: Send DTMF using RFC2833.
Info: Send DTMF using SIP INFO message.
Inband: Send DTMF using inband audio. This requires 64 bit codec,
i.e., PCMU and PCMA.
Auto: Send DTMF using RFC2833 if offered. Otherwise, inband will
be used.
Enable Qualify
If enabled, the UCM6200 will regularly send SIP OPTIONS to the device
to check if the device is still online. The default setting is "No".
Qualify Timeout
When "Enable Qualify" option is set to "Yes", configure the timeout (in ms)
for the Qualify SIP message. If no response is received within the timeout,
the device is considered offline. The default setting is 1000ms.
Qualify Frequency
When "Enable Qualify" 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 Call
Lines
The maximum number of concurrent calls using the trunk. The default
settings 0, which means no limit.
Fax Mode
Select Fax mode. The default setting is “None”.
None: Disable Fax.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
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configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
SRTP
Enable SRTP for the VoIP trunk. The default setting is "No".
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 is allowed to 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.
Table 51: SIP Peer Trunk Configuration Parameters
Basic Settings
Provider Name
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the
trunk.
Transport
Configure the SIP transport protocol to be used in this trunk. The default
setting is "All - UDP Primary".
UDP Only
TCP Only
TLS Only
All - UDP Primary: UDP is the primary transport protocol when all the
other SIP transport methods are available too.
All - TCP Primary: TCP is the primary transport protocol when all the
other SIP transport methods are available too.
All – TLS Primary: TLS is the primary transport protocol when all the
other SIP transport methods are available too.
Keep Original CID
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.
Keep Trunk CID
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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NAT
Turn on this option when the PBX is using public IP and communicating
awith devices behind NAT. If there is one-way audio issue, usually it’s
related to NAT configuration or SIP/RTP port configuration on the firewall.
Disable This Trunk
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.
TEL URI
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-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.
Caller ID
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.
When making outgoing calls, the following rules are used to determine
which CallerID will be used if they exist:
The CallerID configured for the extension will be looked up first.
If no CallerID configured for the extension, the CallerID configured for
the trunk will be used.
If the above two are missing, the "Global Outbound CID" defined in
Web GUI->PBX->Internal Options->General will be used.
CallerID Name
Configure the name of the caller to be displayed when the extension has
no CallerID Name configured.
Auto Record
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 GUI->CDR->Recording Files.
Advanced Settings
Codec Preference
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.263, H.263p and VP8.
DID Mode
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".
DTMF Mode
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 UI->PBX->SIP
Settings->ToS.
RFC2833: Send DTMF using RFC2833.
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Info: Send DTMF using SIP INFO message.
Inband: Send DTMF using inband audio. This requires 64 bit codec,
i.e., PCMU and PCMA.
Auto: Send DTMF using RFC2833 if offered. Otherwise, inband will
be used.
Enable Qualify
If enabled, the UCM6200 will regularly send SIP OPTIONS to the device
to check if the device is still online. The default setting is "No".
Qualify Timeout
When "Enable Qualify" option is set to "Yes", configure the timeout (in ms)
for the Qualify SIP message. If no response is received within the timeout,
the device is considered offline. The default setting is 1000ms.
Qualify Frequency
When "Enable Qualify" 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 Call
Lines
The maximum number of concurrent calls using the trunk. The default
settings 0, which means no limite.
Fax Mode
Select Fax mode. The default setting is “None”.
None: Disable Fax.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
SRTP
Enable SRTP for the VoIP trunk. The default setting is "No".
Sync LDAP Enable
If enabled, the local UCM6200 will automatically provide and update the
local LDAP contacts to the remote UCM6200 SIP peer trunk. In order to
ensure successful synchronization, the remote UCM6200 peer also
needs to enable this option on the SIP peer trunk. The default setting is
"No".
Sync LDAP Password
This is the password used for LDAP contact file encryption and decryption
during the LDAP sync process. The password must be the same on both
UCM6200 peers o ensure successful synchronization.
Sync LDAP Port
Configure the TCP port used LDAP sync feature between two peer
UCM6200.
LDAP Outbound Rule
Specify an outbound rule for LDAP sync feature. The UCM6200 will
automatically modify the remote contacts by adding prefix parsed from
this rule.
LDAP Dialed Prefix
Specify the prefix for LDAP sync feature. The UCM6200 will automatically
modify the remote contacts by adding this prefix.
CC Settings
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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 is allowed to 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.
Table 52: Create New IAX Trunk
Type
Select the VoIP trunk type.
Peer IAX Trunk
Register IAX Trunk
Provider Name
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the
trunk.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
Username
Enter the username to register to the trunk from the provider when
"Register IAX Trunk" type is selected.
Password
Enter the password to register to the trunk from the provider when
"Register IAX Trunk" type is selected.
Disable This Trunk
If selected, the trunk will be disabled.
Table 53: IAX Register Trunk Configuration Parameters
Basic Settings
Provider Name
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the
trunk.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
Disable This Trunk
If selected, the trunk will be disabled.
Caller ID
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
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to set the CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine
which CallerID will be used if they exist:
The CallerID configured for the extension will be looked up first.
If no CallerID configured for the extension, the CallerID configured for
the trunk will be used.
If the above two are missing, the "Global Outbound CID" defined in
Web GUI->PBX->Internal Options->General will be used.
CallerID Name
Configure the name of the caller to be displayed when the extension has
no CallerID Name configured.
Username
Enter the username to register to the trunk from the provider.
Password
Enter the password to register to the trunk from the provider.
Advanced Settings
Codec Preference
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.263, H.263p and VP8.
Enable Qualify
If enabled, the UCM6200 will regularly send SIP OPTIONS to the device
to check if the device is still online. The default setting is "No".
Qualify Timeout
When "Enable Qualify" option is set to "Yes", configure the timeout (in ms)
for the Qualify SIP message. If no response is received within the timeout,
the device is considered offline. The default setting is 1000ms.
Qualify Frequency
When "Enable Qualify" 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 Call
Lines
The maximum number of concurrent calls using the trunk. The default
settings 0, which means no limited.
Fax Mode
Select Fax mode. The default setting is “None”.
None: Disable Fax.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Table 54: IAX Peer Trunk Configuration Parameters
Basic Settings
Provider Name
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
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Host Name
Configure the IP address or URL for the VoIP provider’s server of the
trunk.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
Disable This Trunk
If selected, the trunk will be disabled.
Caller ID
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.
When making outgoing calls, the following rules are used to determine
which CallerID will be used if they exist:
The CallerID configured for the extension will be looked up first.
If no CallerID configured for the extension, the CallerID configured for
the trunk will be used.
If the above two are missing, the "Global Outbound CID" defined in
Web GUI->PBX->Internal Options->General will be used.
CallerID Name
Configure the name of the caller to be displayed when the extension has
no CallerID Name configured.
Advanced Settings
Codec Preference
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.263, H.263p and VP8.
Enable Qualify
If enabled, the UCM6200 will regularly send SIP OPTIONS to the device
to check if the device is still online. The default setting is "No".
Qualify Timeout
When "Enable Qualify" option is set to "Yes", configure the timeout (in ms)
for the Qualify SIP message. If no response is received within the timeout,
the device is considered offline. The default setting is 1000ms.
Qualify Frequency
When "Enable Qualify" 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 Call
Lines
The maximum number of concurrent calls using the trunk. The default
settings 0, which means no limited.
Fax Mode
Select Fax mode. The default setting is “None”.
None: Disable Fax.
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
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DIRECT OUTWARD DIALING (DOD)
The UCM6200 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.
At the moment 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 on how to configure DOD on the UCM6200:
1. To setup DOD go to UCM6200 web GUI->PBX->Basic/Call Routes->VoIP Trunks page.
2. Click to access the DOD options for the selected SIP Trunk.
3. Click "Create a new 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. 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.
Figure 97: DOD extension selection
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6. Click "Save" at the bottom.
Once completed, the user will return to the EDIT DOD page that shows all the extensions that are
associated to a particular DOD.
Figure 98: Edit DOD
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SLA STATION
The UCM6200 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 UCM6200 is similar to BLF but SLA is used to monitor external line i.e.,
analog trunk on the UCM6200. 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 UCM6200.
CREATE/EDIT SLA STATION
SLA Station can be configured on web GUI->PBX->Basic/Call Routes->SLA Station.
Figure 99: SLA Station
Click on “Create New SLA Station” to add a 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 55: 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.
Selected 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
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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.
Ring Delay
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.
Hold Access
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 UCM6200, go to web UI->Basic/Call Routes->Analog 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 UI->Basic/Call
Routes->SLA Station page.
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Figure 100: Enable SLA Mode for Analog Trunk
Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.
Figure 101: Analog Trunk with SLA Mode Enabled
2. On the UCM6200, go to web UI->Basic/Call Routes->SLA Station page, click on “Create New SLA
Station”. 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 102: SLA Example - SLA Station
3. On the SIP phone 1, configure to register UCM6200 extension 1002. Configure the MPK as BLF mode
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and the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case.
4. On the SIP phone 2, configure to register UCM6200 extension 1005. Configure the MPK as BLF mode
and value must be set to “extension_trunkname”, which is 1005_fxo1 in this case.
Figure 103: 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 UCM6200, 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 a 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 GUI->PBX->Basic/Call Routes->Outbound Routes to add and edit outbound rules.
Click on "Create New Outbound Rule" to add a new outbound route.
Click on to edit the outbound route.
Click on to delete the outbound route.
On the UCM6200, the outbound route priority is based on “Best matching pattern”. For example, the
UCM6200 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, users can click on to move the outbound route up/down to
arrange the priority among those outbound routes.
Table 56: Outbound Route Configuration Parameters
Calling Rule Name
Configure the name of the calling rule (e.g., local, long_distance, and etc).
Letters, digits, _ and - are allowed.
Pattern
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.
Password
Configure the password for users to use this rule when making outbound
calls.
Call Duration Limit
Enable to configure the maximum duration for the call using this outbound
route.
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Maximum Call Duration
Configure the maximum duration of the call (in seconds). The default
setting is 0, which means no limit.
Warning Time
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.
Warning Repeat Interval
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.
Privilege Level
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 are allowed to
use this rule.
National: Users with National or International level are allowed to 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.
Enable Filter on Source
Caller ID
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.
1. Select available extensions/extension groups from the left to the right.
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.
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Example: [12345-9] - Any digit from 1 to 9.
Send This Call Through Trunk
Use Trunk
Select the trunk for this outbound rule.
Strip
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.
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.
Use Failover Trunk
Failover Trunk
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.
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.
Strip
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.
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.
INBOUND ROUTES
Inbound routes can be configured via Web GUI->PBX->Basic/Call Routes->Inbound Routes.
Click on "Create New Inbound Rule" to add a new inbound route.
Click on "Blacklist" to configure blacklist for all inbound routes.
Click on to edit the inbound route.
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Click on to delete the inbound route.
INBOUND RULE CONFIGURATIONS
Table 57: Inbound Rule Configuration Parameters
Trunks
Select the trunk to configure the inbound rule.
DID Pattern
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, divided by a ‘/’ character.
The first part is used to specify the dialed number 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 is allowed to call in or call out.
For example, patter '_2XXX/1234' means the only extension with the
caller ID '1234' is allowed to use this rule.
Prepend Trunk Name
Prepend trunk name to display
Alert-Info
Configure the Alert-Info, when UCM6200 receives an INVITE request, the
Alert-Info header field specifies an alternative ring tone to the UAS.
Inbound Multiple Mode
Multiple mode allows user to switch between destinations of the inbound
rule by feature codes. Configure related feature codes in the “Feature
Codes” page. If this option is enabled, user can use feature code to
switch between different destinations.
Default Destination
Select the default destination for the inbound call.
Extension
Voicemail
Conference Room
Queue
Ring Group
Paging/Intercom
Voicemail Group
Fax
DISA
IVR
Dial By Name
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External Number
By DID
When "By DID" is used, the UCM6200 will look for the destination
based on the number dialed, which could be local extensions,
conference, call queue, ring group, paging/intercom group, IVR,
voicemail groups and Fax extension as configured in "DID
destination". If the dialed number matches the DID pattern, the call
will be allowed to go through.
Strip
Configure the number of digits to be stripped from the beginning of the
DID. This option shows up only when "By DID" is selected.
Prepend
Configure the number of digits to be prepended to an inbound DID
pattern, with strip taking precedence over prepend.
Dial Trunk
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 UCM6200’s trunk.
DID Destination
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
Fax Extension
Dial By Name
All
Time Condition
Time Conditions
Select the time condition for the inbound rule.
Destination
Select the destination for the inbound call during the specified time
condition.
INBOUND ROUTE: PREPEND EXAMPLE
UCM6200 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 in order to route calls to different extensions.
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Figure 104: Inbound Route feature: Prepend
The following example demonstrates the process,
1. If Trunk provides a DID pattern of 18005251163.
2. If Strip is set to 8, UCM6200 will strip the first 8 digits.
3. If Prepend is set to 2, UCM6200 will then prepend a 2 to the stripped number, now the number
become 2163.
4. UCM6200 will now forward the incoming call to extension 2163.
INBOUND ROUTE: MULTIPLE MODE
In the UCM6200, 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 105: 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 this route. By default, the call coming into this inbound route will be routed to the default
destination.
SIP end devices that have registered on the UCM6200 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 UI 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.
FAX INTELLIGENT ROUTE
The UCM6200 can automatically detect Fax and phone signal coming from the FXO port, and then forward Fax
or phone signal to the right destination. For example, when a regular phone call is coming, the UCM6200 will be
able to detect the phone signal and forward it through the correct inbound route to the destination; if Fax signal is
coming, the UCM6200 will be able to forward it to the FXS extension where the Fax machine is connected.
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FAX WITH TWO MEDIA
The UCM6200 supports Fax re-invite with multiple codec negotiation. If a Fax re-invite contains both T.38 and
PCMA/PCMU codec, UCM6200 will choose T.38 codec over PCMA/PCMU.
BLACKLIST CONFIGURATIONS
In the UCM6200, 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 to add to the list.
To remove a number from the Blacklist, select the number in "Blacklist list" and click on .
Figure 106: Blacklist Configuration Parameters
To add blacklist number in batch, click on to upload blacklist file in csv format. The supported csv
format is as below.
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Figure 107: 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 GUI->PBX->Internal Options->Feature Codes.
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CONFERENCE BRIDGE
The UCM6200 supports conference bridge allowing multiple bridges used at the same time:
UCM6202/6204 supports up to 3 conference bridges allowing up to 25 simultaneous PSTN or IP
participants.
UCM6208 supports up to 6 conference bridges allowing up to 32 simultaneous PSTN or IP participants.
The conference bridge configurations can be accessed under Web GUI->PBX->Call
Features->Conference. In this page, users could create, edit, view, invite, manage the participants and
delete conference bridges. The conference bridge status and conference call recordings (if recording is
enabled) will be displayed in this web page as well.
CONFERENCE BRIDGE CONFIGURATIONS
Click on "Create New Conference Room" to add a new conference bridge.
Click on to edit the conference bridge.
Click on to delete the conference bridge.
Table 58: Conference Bridge Configuration Parameters
Extension
Configure the conference number for the users to dial into the
conference.
Password
When configured, the users who would like to join the conference call
must enter this password before accessing the conference bridge.
Note:
If "Public Mode" is enabled, the password is not required to join the
conference bridge thus this field is invalid.
The password has to be at least 4 characters.
Admin Password
Configure the password to join the conference bridge as administrator.
Conference administrator 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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Note:
If "Public Mode" is enabled, the password is not required to join the
conference bridge thus this field is invalid.
The password has to be at least 4 characters.
Enable Caller Menu
If enabled, conference participant could press the * key to access the
conference bridge menu. The default setting is "No".
Record Conference
If enabled, the calls in this conference bridge 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".
Quiet Mode
If enabled, if there are users joining or leaving the conference, voice
prompt or notification tone won't be played. The default setting is "No".
Note:
"Quiet Mode" and "Announce Callers" cannot be enabled at the same
time.
Wait For Admin
If enabled, the participants will not hear each other until the conference
administrator 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.
Enable 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
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.
Public Mode
If enabled, no authentication will be required when joining the conference
call. The default setting is "Yes".
Play Hold Music
If enabled, the UCM6200 will play Hold music when there is only one user
in the conference. The default setting is "No".
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Music On Hold
Select the music on hold class to be played in conference call. Music On
Hold class can be set up under web UI->PBX->Internal Options->Music
On Hold.
Skip Authentication When
Inviting User via Trunk from
Web GUI
If enabled, the invitation from Web GUI for a conference bridge with
password will skip the authentication for the invited users. The default
setting is "No".
JOIN A CONFERENCE CALL
Users could dial the conference bridge 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 UCM6200 conference bridge, there are two ways to invite other parties to join the
conference.
Invite from Web GUI.
For each conference bridge in UCM6200 Web GUI->PBX->Call Features->Conference, 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 108: Conference Invitation From Web GUI
Invite by dialing 0 or 1 during conference call.
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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 bridge 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 bridge as administrator, enter the admin password when joining the conference. A
conference bridge can have multiple administrators.
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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 UCM6200 web GUI during the conference call, the participants in each conference bridge 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 bridge so that other users cannot join it anymore.
4. Click on to invite other users into the conference bridge.
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.
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Table 59: Conference Caller IVR Menu
Conference Administrator IVR Menu
1
Mute/unmute yourself.
2
Lock/unlock the conference bridge.
3
Kick the last joined user from the conference.
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
More options.
1: List all users currently in the conference call.
2: Kick all non-Administrator participants from the conference call.
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 bridge configuration cannot be modified.
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RECORD CONFERENCE
The UCM6200 allows users to record the conference call and retrieve the recording from web
GUI->PBX->Call Features->Conference.
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To record the conference call, when the conference bridge is in idle, enable "Record Conference" from the
conference bridge 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 Selected
Recording Files” after selecting the recording files.
Figure 109: Conference Recording
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CONFERENCE SCHEDULE
CONFERENCE SCHEUDLE CONFIGURATION
Conference Schedule can be found under UCM6200 web UI->PBX->Call Features->Conference Schedule.
Users can create, edit, view and delete a Conference Schedule.
Click on “Create New Conference Schedule” to add a new Conference Schedule.
Click on the scheduled conference to edit or delete the event.
After the user configures UCM6200 with Google Service Settings [GOOGLE SERVICE SETTINGS SUPPORT]
and enables Google Calendar for Conference Schedule, the conference schedule on the UCM6200 can be
synchronized with Google Calendar for authorized Google account.
Table 60: Conference Schedule Parameters
Schedule Options
Conference Topic
Configure the name of the scheduled conference. Letters, digits, _ and -
are allowed.
Conference Room
Select a conference room for this scheduled conference.
Kick Time(m)
Set kick time before conference starts. When kick time is reached, a
warning prompt will be played for all attendees in the conference room.
After 5 minutes, this conference room will be cleared and locked for the
scheduled conference to begin.
Note: Kick Time cannot be less than 6 minutes in order to clear the
conference room.
Description
The description of scheduled conference.
Repeat
Repeat interval of scheduled conference. By default it’s set to single
event.
Schedule Time
Configure the beginning date and duration of scheduled conference.
Note: Please pay attention to avoid time conflict on schedules in the
same conference room.
Enable Google Calendar
Select this option to sync scheduled conference with Google Calendar.
Note: Google Service Setting OAuth2.0 must be configured on the
UCM6200. Please refer to section [GOOGLE SERVICE SETTINGS
SUPPORT].
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Conference Administrator
Select the administrator of scheduled conference from selected
extensions.
Note:
“Public Mode” must be disabled from Conference Room Options tab.
Local Extension
Select available extensions from the list to attend scheduled conference.
Remote Extension
Select available extensions from the remote peer PBX.
Note: “LDAP Sync” must be enabled on the UCM6200 in order to view
remote extensions here.
Special Extension
Add extensions that are not in the list (both local and remote list). If the
user wishes to add the special extension, please match the pattern on the
outbound route.
Remote Conference
Invite a remote conference.
Conference Room Options
Password
Configure conference room password. Please note that if “Public Mode” is
enabled, this option is automatically disabled.
Admin Password
Configure the password to join as conference administrator. Please note
that if “Public Mode” is enabled, this option is automatically disabled.
Enable Caller Menu
If this option is enabled, conference participants will be able to access
conference bridge menu by pressing the * key.
Record Conference
If this option is enabled, conference call will be recorded in .wav format.
The recorded file can be found from Conference page.
Quiet Mode
If this option is enabled, the notification tone or voice prompt for joining or
leaving the conference won’t be played.
Note: Option “Quiet Mode” and option “Announce Caller” cannot be
enabled at the same time.
Wait For Admin
If this option is enabled, the participants in the conference won’t be able to
hear each other until conference administrator joins the conference.
Note: If “Quiet Mode” is enabled, voice prompt for this option won’t be
played.
Enable User Invite
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 bridge to another conference
room
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Press ‘3’ to drop all current multi-conference bridges
Note: Conference Administrator is always allowed to access this menu.
Announce Callers
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.
Public Mode
If this option is enabled, no authentication is required for entering the
conference room.
Note: Please be aware of the potential security risks when turning on this
option.
Play Hold Music
If this option is enabled, UCM6200 will play Hold Music while there is only
one participant in the conference room or the conference is not yet
started.
Skip Authentication When
Inviting Users via Trunk from
Web GUI
If this option is enabled, the invitation from Web GUI via a trunk with
password won’t require authentication.
Note: Please be aware of the potential security risks when turning on this
option.
Cleaner Options
Cleaner Options
Enable Conference
Schedules Cleaner
If this option is enabled, conference schedules will be automatically
cleaned as configured.
Conference Schedules Clean
Time
Enter the clean time (in hours). The valid range is from 0 to 23.
Clean Interval
Enter the clean interval (in days). The valid range is from 1 to 30.
Show/hide Conference Schedule Table
Enable this option will allow web UI to display scheduled conference in Conference Schedule Table. Please
see figure below.
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Figure 110: Conference Schedule
Once the conference room is scheduled, at the kick time, all users will be removed from conference room
and no extension is allowed to join the conference room anymore. At the scheduled conference time,
UCM6200 will send INVITE to the extensions that have been selected for conference.
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Note:
Please make sure that outbound route is properly configured for remote extensions to join the
conference.
Once Kick Time is reached, Conference Schedule is locked and cannot be modified.
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IVR
CONFIGURE IVR
IVR configurations can be accessed under the UCM6200 Web GUI->PBX->Call Features->IVR. Users
could create, edit, view and delete an IVR.
Click on "Create New IVR" to add a new IVR.
Click on to edit the IVR configuration.
Click on to delete the IVR.
Table 61: IVR Configuration Parameters
Basic Settings
Name
Configure the name of the IVR. Letters, digits, _ and - are allowed.
Extension
Enter the extension number for users to access the IVR.
DID Destination
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
Voicemail Groups
Fax Extension
Dial By Name
All
Dial Trunk
If enabled, all callers to the IVR is allowed to use trunk. The permission
must be configured for the users to use the trunk first. The default setting
is "No".
Permission
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 UCM6200 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
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allowed to go through.
Welcome Prompt
Select an audio file to play as the welcome prompt for the IVR. Click on
"Prompt" to add additional audio file under web GUI->Internal
Options->IVR 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 UCM6200 will consider the entries
complete. The default timeout is 3 seconds.
Response Timeout
After playing the prompts in the IVR, the UCM6200 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 Prompt
Select the prompt message to be played when an invalid extension is
pressed.
Response Timeout Repeat
Loops
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 Repeat Loops
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 GUI->PBX->Internal Options->Language. The dropdown list
shows all the current available voice prompt languages on the UCM6200.
To add more languages in the list, please download voice prompt
package by selecting "Check Prompt List" under web
GUI->PBX->Internal Options->Language.
Key Pressing Events
Key Press Event:
Press 0
Press 1
Press 2
Press 3
Press 4
Press 5
Press 6
Press 7
Press 8
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
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Press 9
Press *
Timeout
Invalid
Fax
Custom Prompt
Hangup
DISA
Dial By Name
External Number
Callback
CREATE CUSTOM PROMPT
To record new IVR prompt or upload IVR prompt to be used in IVR, click on “Prompt” next to the “Welcome
Prompt” option and the users will be redirected to Custom Prompt page. Or users could go to Web
GUI->PBX->Internal Options->Custom Prompt page directly.
Figure 111: Click on Prompt to Create IVR Prompt
Once the IVR prompt file is successfully added to the UCM6200, it will be added into the prompt list
options for users to select in different IVR scenarios.
RECORD NEW CUSTOM PROMPT
Settings
In the UCM6200 web UI->PBX->Internal Options->Custom Prompt page, click on “Record New Custom
Prompt” and follow the steps below to record new IVR prompt.
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Figure 112: Record New Custom Prompt
Specify the IVR file name.
Select the format (GSM or WAV) for the IVR prompt file to be recorded.
Select the extension to receive the call from the UCM6200 to record the IVR prompt.
Click the “Record” button. A request will be sent to the UCM6200. The UCM6200 will then call the
extension for recording the IVR prompt from the phone.
Pick up the call from the extension and start the recording following the voice prompt.
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 Custom Prompt” in Web
GUI->PBX->Internal Options->Custom Prompt page to upload the file to the UCM6200. The following
are required for the IVR prompt file to be successfully uploaded and used by the UCM6200:
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.
Figure 113: Upload Custom Prompt
Click on to select audio file from local PC and click on to start uploading. Once uploaded, the file
will appear in the Custom Prompt web page.
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LANGUAGE SETTINGS FOR VOICE PROMPT
The UCM6200 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, Swedish and Turkish.
English (United States) and Chinese voice prompts are built in with the UCM6200 already. The other
languages provided by Grandstream can be downloaded and installed from the UCM6200 web GUI
directly. Additionally, users could customize their own voice prompts, package them and upload to the
UCM6200.
Language settings for voice prompt can be accessed under Web GUI->PBX->Internal
Options->Language.
DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE
To download and install voice prompt package in different languages from UCM6200 web GUI, click on
"Check Prompt List" button.
Figure 114: Language Settings for Voice Prompt
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.
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Figure 115: Voice Prompt Package List
Click on to download the language to the UCM6200. The installation will be automatically started
once the downloading is finished.
Figure 116: New Voice Prompt Language Added
A new language option will be displayed after successfully installed. Users then could select it to apply in
the UCM6200 system voice prompt or delete it from the UCM6200.
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CUSTOMIZE SPECIFIC PROMPT
On the UCM6200, if the user needs to replace some specific customized prompt, the user can upload a single
specific customized prompt from web UI->PBX->Internal Options->Language instead of the entire language
pack.
Figure 117: Upload Single Voice Prompt for Entire Language Pack
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VOICEMAIL
CONFIGURE VOICEMAIL
If the voicemail is enabled for UCM6200 extensions, the configurations of the voicemail can be globally set
up and managed under Web GUI->PBX->Call Features->Voicemail.
Table 62: Voicemail Settings
Max Greeting
Configure the maximum number of seconds for the voicemail greeting.
The default setting is 60 seconds.
Dial ‘0’ For Operator
If enabled, the caller can press 0 to exit the voicemail application and
connect to the configured operator’s extension. The operator extension
can be configured under web GUI->PBX->Internal Options->General.
Max Messages Per Folder
Configure the maximum number of messages per folder in users’
voicemail. The valid range 10 to 1000. The default setting is 50.
Max Message Time
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
Min Effective Message Time
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.
Announce Message Caller-ID
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
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is "No".
Announce Message Duration
If enabled, the message duration will be announced at the beginning of
the voicemail message. The default setting is "No".
Play Envelope
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".
Play from Last
If enabled, UCM will play from the voice message left most recently; if
disabled, UCM will play from the earliest left voice message
Allow User Review
If enabled, users can review the message following the IVR before
sending the message out. The default setting is "No".
ACCESS VOICEMAIL
If the voicemail is enabled for UCM6200 extensions, the users can dial the voicemail access feature code
(by default *98 or *97) to access the extension’s voicemail. The users will be prompt to enter the voicemail
password and then can enter digits from the phone keypad to navigate in the IVR menu for different
options.
Table 63: Voicemail IVR Menu
Main Menu
Sub Menu 1
Sub Menu 2
1 - New
messages
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
7 - Delete this message
8 - Forward the message to another user
9 – Save
* - Help
# - Exit
2 - Change
folders
0 - New messages
1 - Old messages
2 - Work messages
3 - Family messages
4 - Friend messages
# - Cancel
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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
0 - Mailbox
options
1 - Record your unavailable message
1 - Accept this recording
2 - Listen to it
3 - Re-record your message
2 - Record your busy message
1 - Accept this recording
2 - Listen to it
3 - Re-record your message
3 - Record your name
1 - Accept this recording
2 - Listen to it
3 - Re-record your message
4 - Record temporary greeting
1 - Accept this recording
2 - Listen to it
3 - Re-record your message
5 - Change your password
* - Return to the main menu
VOICEMAIL EMAIL SETTINGS
The UCM6200 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 64: Voicemail Email Settings
Attach Recordings to E-Mail
If enabled, voicemails will be sent to user's Email address. The default
setting is "Yes".
Keep Recordings
If enabled, voicemail will be stored in the UCM6200 after the email is sent.
The default setting is “Yes”.
Template For Voicemail Emails
Fill in the "Subject:" and "Message:" content, to be used in the Email
when sending to the user.
The template variables are:
\t: TAB
${VM_NAME}: Recipient's first name and last name
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${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
Figure 118: Voicemail Email Settings
Click on "Load Default Settings" button to view the default template as an example.
CONFIGURE VOICEMAIL GROUP
The UCM6200 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->PBX->Call
Features->Voicemail Group. Click on "Create New Voicemail Group" to configure the group.
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Figure 119: Voicemail Group
Table 65: Voicemail Group Settings
Extension
Enter the Voicemail Group Extension. The voicemail messages left to this
extension will be forwarded to all the voicemail group members.
Name
Configure the Name to identify the voicemail group. Letters, digits, _ and -
are allowed.
Voicemail Password
Configure the voicemail password for the users to check voicemail
messages.
Email Address
Configure the Email address for the voicemail group extension.
Voicemail Group Mailboxes
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 UCM6200 supports ring group feature with different ring strategies applied to the ring group members.
This section describes the ring group configuration on the UCM6200.
CONFIGURE RING GROUP
Ring group settings can be accessed via Web GUI->PBX->Call Features->Ring Group.
Figure 120: Ring Group
Click on “Create New Ring Group” 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.
Table 66: Ring Group Parameters
Ring Group Name
Configure ring group name to identify the ring group. Letters, digits, _ and
– are allowed.
Extension
Configure the ring group extension.
Ring Group Members
Select available users from the left side to the ring group member list on
the right side. Click on to arrange the order.
Selected LDAP Numbers
Select available remote users from the left side to the ring group member
list on the right side. Click on to arrange the order. Note:
LDAP Sync must be enabled first.
Ring Strategy
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 ringing.
Ring in order.
Ring the members with the order configured in ring group list. If the
first member doesn’t answer the call, it will stop ringing the first
member and start ringing the second member.
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Custom Prompt
This option is to set a custom prompt for a ring group to announce to
caller. Click on ‘Prompt’, it will direct to the page PBX->Internal
Options->Custom Prompt, where users could record new prompt or
upload prompt files.
Ring Timeout on Each Member
Configure the number of seconds to ring each member. If set to 0, it will
keep ringing. The default setting is 30 seconds.
Note:
The actual ring timeout might be overridden by users if the phone has ring
timeout settings as well.
Auto Record
If enabled, calls on this ring group will be automatically recorded. The
default setting is No. The recording files can be accessed from web
GUI->CDR->Recording Files.
Enable Destination
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.
Secret
Configure the password to access the ring group extension's voicemail.
Note:
The password has to be at least 4 characters.
Email Address
Configure the Email address of the ring group extension's voicemail. If
"Attach Recordings to E-mail" is enabled from Web
GUI->PBX->Voicemail->Voicemail Email Settings, the voicemail can
be sent to the ring group's Email address as attachment.
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Figure 121: Ring Group Configuration
REMOTE EXTENSION IN RING GROUP
Remote extensions from the peer trunk of a remote UCM6200 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 UCM6200_A and UCM6200_B. SIP Trunk can be found under
web UI-> PBX-> Basic/Call Routes-> VoIP 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 UCM6200_A update remote LDAP server automatically from peer UCM6200_B. In
addition, Sync LDAP Password must match for UCM6200_A and UCM6200_B in order to sync LDAP
contact automatically. Port number can be anything between 0~65535, and use the outbound rule
created in step 1 for the LDAP Outbound Rule option.
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Figure 122: Sync LDAP Server option
3. In case if LDAP server doesn’t 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 UCM6200.
Figure 123: Manually Sync LDAP Server
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4. Under Ring Groups setting page, click . Ring Groups can be found
under web UI-> PBX-> Call Features-> Ring 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 UCM6200 can be included into that UCM6200’s LDAP contact.
Figure 124: 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 UCM6200 paging and intercom can
be used via feature code to a single extension or a paging/intercom group. This sections describes the
configuration of paging/intercom group under Web GUI->PBX->Call Features->Paging/Intercom.
CONFIGURE PAGING/INTERCOM GROUP
Click on "Create New Paging/Intercom Group" to add paging/intercom group.
Figure 125: Paging/Intercom Group
Table 67: Paging/Intercom Group Configuration Parameters
Name
Configure paging/intercom group name.
Extension
Configure the paging/intercom group extension.
Type
Select "2-way Intercom" or "1-way Page".
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 to the page
PBX->Internal Options->Custom Prompt, where users could record
new prompt or upload prompt files.
Page/Intercom Group
Members
Select available users from the left side to the paging/intercom group
member list on the right.
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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.
Figure 126: Page/Intercom Group Settings
The UCM6200 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 GUI->PBX->Internal
Options->Feature Codes directly.
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CALL QUEUE
The UCM6200 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 GUI->PBX->Call
Features->Call Queue.
CONFIGURE CALL QUEUE
Call queue settings can be accessed via Web GUI->PBX->Call Features->Call Queue.
Figure 127: Call Queue
Click on "Create New Queue" to add call queue.
Click on to edit the call queue. The call queue configuration parameters are listed in the table
below.
Table 68: Call Queue Configuration Parameters
Extension
Configure the call queue extension.
Name
Configure the call queue name to identify the call queue.
Strategy
Select the strategy for the call queue.
Ring All
Ring all available Agents simultaneously until one answers.
Linear
Ring agents in the specified order.
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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".
Music On Hold
Select the Music On Hold class for the call queue.
Note:
Music On Hold classes can be managed from Web GUI-> PBX->Internal
Options->Music On Hold.
Leave When Empty
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.
Dial in Empty Queue
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.
Strict
Callers cannot dial into a queue if the agents are paused, invalid or
unavailable.
Dynamic Login Password
If enabled, the configured PIN number is required for dynamic agent to
log in. The default setting is disabled.
Ring Time Out
Configure the number of seconds an agent will ring before the call goes to
the next agent. The default setting is 15 seconds.
Wrapup Time
Configure the number of seconds before a new call can ring the queue
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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 15 seconds.
Max Queue Length
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.
Report Hold Time
If enabled, the UCM6200 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".
Wait Time
If enabled, users will be disconnected after the configured number of
seconds. The default setting is "No".
Note:
It is recommended to configure "Wait Time" longer than the "Wrapup
Time".
Auto Record
If enabled, the calls on the call queue will be automatically recorded. The
recording files can be accessed in Queue Recordings under web
GUI->PBX->Call Features->Call Queue.
Enable Destination
If enabled, the incoming call for the call queue will be routed to the
destination configured in the next field if none of the agents answers the
call after ringing for a time of “Ring Timeout”.
Queue Timeout
Configure the global timeout (in seconds) of call queue. It must be bigger
than the value of ring timeout. The call in the queue will be transferred to
the failover destination directly if this time is exceeded.
Failover Destination
Configure the call destination for the call to be routed to if no agent in this
call queue answers the call.
Enable Feature Codes
Enable feature codes option for call queue. For example, *83 is used for
“Agent Pause”
Agents
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 arrange the order.
Click on to delete the call queue.
Click on "Agent Login Settings" to configure Agent Login Extension Postfix and Agent Logout
Extension Postfix. Once configured, users could log in the call queue as dynamic agent.
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Figure 128: 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 doesn't 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
GUI->PBX->Internal Options->Feature Codes. The default feature code is *83 for "Agent Pause"
and *84 for "Agent Unpause".
Queue recordings are shown on the Call Queue page. Click on to download the recording file
in .wav format; click on to delete the recording file. To delete multiple 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.
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EXTENSION GROUPS
The UCM6200 extension group feature allows users to assign and categorize extensions in different
groups to better manage the configurations on the UCM6200. 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 GUI->PBX->Call Features->Extension Groups.
Click on "Create New Extension Group" to create a new extension group.
Click on to edit the extension group.
Select extensions from the list on the left side to the right side.
Figure 129: Edit Extension Group
Click on to delete the extension group.
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USING EXTENSION GROUPS
Here is an example where the extension group can be used. Go to Web GUI->PBX->Basic/Call
Routes->Outbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and
extension groups will show up for users to select.
Figure 130: Select Extension Group in Outbound Route
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PICKUP GROUPS
The UCM6200 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 GUI->PBX->Call Features->Pickup Groups.
Click on "Create New Pickup Group" to create a new pickup group.
Click on to edit the pickup group.
Select extensions from the list on the left side to the right side.
Figure 131: Edit Pickup Group
Click on to delete the 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’s not necessary to add the extension number after the pickup feature code. The pickup feature code is
configurable under Web GUI->PBX->Internal Options->Feature Codes.
The default pickup feature code is *8.
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Figure 132: Edit Pickup Feature Code
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MUSIC ON HOLD
Music On Hold settings can be accessed via Web GUI->PBX->Internal Options->Music 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 133: Music On Hold Default Class
Click on "Create New MOH Class" to add a new Music On Hold class.
Click on to configure the MOH class sort method to be "Alpha" or "Random" for the sound files.
Click on next to the selected Music On Hold class to delete this Music On Hold class.
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Click on to select music file from local PC and click on to start uploading. The music file
uploaded has to be 8 KHz Mono format with size smaller than 5M.
Click on next to the sound file to delete it from the selected Music On Hold Class.
Select the sound files and click on to delete all selected music on
hold files.
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Note:
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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FAX/T.38
The UCM6200 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
GUI->PBX->Internal Options->FAX/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.
Table 69: 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.
Minimum Transfer Rate
Configure the minimum transfer rate during the Fax rate negotiation. The
possible values are 2400, 4800, 7200, 9600, 12000 and 14000. The
default setting is 2400.
Max Concurrent Sending Fax
Configure the concurrent fax that can be sent by UCM6200. Two mode
“Only” and “More” are supported.
Only
Under this mode, the UCM6200 allows only single user to send fax at
a time.
More
Under this mode, the UCM6200 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.
Default Email Address
Configure the Email address to send the received Fax to if user's Email
address cannot be found.
Note:
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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 be not be received from Email.
Template Variables
Fill in the "Subject:" and "Message:" content, to be used in the Email
when sending the Fax to the users.
The template variables are:
${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
Click on to edit the Fax extension.
Click on to delete the Fax extension.
SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE
The following instructions describe how to use the UCM6200 to receive Fax from PSTN line on the Fax
machine connected to the UCM6200 FXS port.
1. Connect Fax machine to the UCM6200 FXS port.
2. Connect PSTN line to the UCM6200 FXO port.
3. Go to web GUI->PBX->Analog Trunks page.
4. Create or edit the analog trunk for Fax as below.
Fax Mode: Make sure "Fax Mode" option is set to "None".
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Figure 134: Configure Analog Trunk without Fax Detection
5. Go to UCM6200 web GUI->PBX->Basic/Call Routes->Extensions page.
6. Create or edit the extension for FXS port.
Analog Station: Select FXS port to be assigned to the extension. By default, it's set to "None".
Once selected, analog related settings for this extension will show up in "Analog Settings"
section.
Figure 135: Configure Extension for Fax Machine: FXS Extension
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Figure 136: Configure Extension for Fax Machine: Analog Settings
7. Go to web GUI->PBX->Basic/Call Routes->Inbound 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 137: Configure Inbound Rule for Fax
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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.
SAMPLE CONFIGURATION FOR FAX-TO-EMAIL
The following instructions describe a sample configuration on how to use Fax-to-Email feature on the
UCM6200.
1. Connect PSTN line to the UCM6200 FXO port.
2. Go to UCM6200 web GUI->Internal Options->Fax/T.38 page. Create a new Fax extension.
Figure 138: Create Fax Extension
3. Go to UCM6200 web GUI->Basic/Call Routes->Analog Trunks page. Create a new analog trunk.
Please make sure "Fax Detection" is set to "No".
4. Go to UCM6200 web GUI->Basic/Call Routes->Inbound Routes page. Create a new inbound route
and set the default destination to the Fax extension.
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Figure 139: 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 file and sent to the Email address Faxtest@ucm6200mycompany.com as
attachment.
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ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)
The UCM6200 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’s
particularly useful when the system admin tries to track the state of a telephony client inside Asterisk.
User could configure AMI parameters on UCM6200 web GUI->PBX->Internal Options->AMI. For details
on how to use AMI on UCM6200, please refer to the following AMI guide:
http://www.grandstream.com/sites/default/files/Resources/ucm6200_AMI_guide.pdf
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Warning:
Please do not enable AMI on the UCM6200 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 UCM6200 system. Please be cautious when enabling AMI access on the UCM6200 and restrict the
permission granted to the AMI user. By using AMI on UCM6200 you agree you understand and
acknowledge the risks associated with this.
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BUSY CAMP-ON
The UCM6200 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/ucm6200_busy_camp_on_guide.pdf
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FOLLOW ME
Follow Me is a feature on the UCM6200 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 web GUI-> PBX-> Call Features->Follow Me.
To configure follow me:
Click on "Create New Follow Me" and then select an extension to be configured with Follow Me.
Figure 140: Create Follow Me
Click on “Next” to continue editing Follow Me configuration.
Figure 141: Edit Follow Me
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Click on “Add Follow Me Number” to add local extensions or external numbers to be called after
ringing the extension selected in the first step.
Once created, it will be displayed on the follow me web page list. Click on to edit the Follow Me
configuration. Click on to delete the Follow Me.
The following table shows the Follow Me configuration parameters.
Table 70: Follow Me Settings
Enable
Configure to enable or disable Follow Me for this user.
Skip Trunk Auth
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.
Music On Hold Class
Configure the Music On Hold class that the caller would hear while
tracking the user.
Confirm When Answering
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 it.
Enable Destination
When enabled, the call will be routed to the default destination if no one in
the Follow Me extensions answers the call.
Default Destination
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
Follow Me Numbers
The added numbers are listed here. Click on to arrange the
order. Click on to delete the number. Click on to add new
numbers.
New Follow Me Number
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.
Dialing Order
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.
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Click on “Follow Me Options” to enable or disable the options listed in the following table.
Table 71: Follow Me Options
Playback Incoming Status
Message
If enabled, the PBX will playback the incoming status message before
starting the Follow Me steps.
Record the Caller’s Name
If enabled, the PBX will record the caller’s name from the phone so it can
be announced to the callee in each step.
Playback Unreachable Status
Message
If enabled, the PBX will playback the unreachable status message to the
caller if the callee cannot be reached.
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ONE-KEY DIAL
The UCM6200 supports One-Key Dial that allows users to call a certain destination by pressing one digit 0
to 9 on the keypad. This creates a system-wide speed dial access for all the extensions on the UCM6200.
To enable One-Key Dial, on the UCM6200 web GUI, go to page PBX->Call Features->One-Key Dial.
Figure 142: Configure One-Key Dial
User should first decide a digit used for One-Key Dial and check the option “Enable Destination” for the
digit. Then select a dial destination from “Default Destination”. The supported destinations include
extension, voicemail, conference room, voicemail group, IVR, ring group, call queue, page group, fax,
DISA, Dial by Name and external number.
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Figure 143: One-Key Dial Destinations
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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’s using his cell phone, pay phone, regular PSTN, etc. After calling into
UCM6200, the user can then dial out via the SIP trunk or PSTN trunk connected to UCM6200 as it is an
internal extension.
The UCM6200 supports DISA to be used in IVR or inbound route. Before using it, create new DISA under
web GUI->Call Features->DISA.
Click on "Create New IVR" to add a new DISA.
Click on to edit the DISA configuration.
Click on to delete the DISA.
Figure 144: Create New DISA
Table 72: DISA Settings
Name
Configure DISA name to identify the DISA.
Password
Configure the password (digit only) required for the user to enter before
using DISA to dial out.
Note:
The password has to be at least 4 digits.
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Permission
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 UCM6200 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.
Response Timeout
Configure the maximum amount of time the UCM6200 will wait before
hanging up if the user dials an incomplete or invalid number. The default
setting is 10 seconds.
Digit Timeout
Configure the maximum amount of time permitted between digits when
the user is typing the extension. The default setting is 5 seconds.
Allow Hangup
If enabled, during an active call, users can enter the UCM6200 hangup
feature code (by default it's *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".
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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CALLBACK FEATURE
Callback is mainly 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 UCM6200.
2. On the UCM6200, 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 UCM6200 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 UCM6200 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 UCM6200 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 UCM6200, go to web GUI->PBX->Call Features->Callback page and click
on . Configuration parameters are listed in the following table.
Table 73: Callback Configuration Parameters
Name
Configure a name to identify the Callback.
CallerID Pattern
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.
Outbound Prepend
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.
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Delay Before Callback
Configure the number of seconds to be delayed before calling back the
user.
Destination
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 UCM6200
trunk.
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BLF AND EVENT LIST
BLF
The UCM6200 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).
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EVENT LIST
Besides BLF, users can also configure the phones to monitor event list. In this way, both local extensions
on the same UCM6200 and remote extensions on the VOIP trunk can be monitored. The event list setting
is under web GUI->Call Features->Event List.
Click on "Create New Event List" to add a new event list.
Click on to edit the event list configuration.
Click on to delete the event list.
Table 74: Event List Settings
URI
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 UCM6200. The valid characters are letters, digits, _ and -.
Local Extensions
Select the available extensions/Extension Groups listed on the local
UCM6200 to be monitored in the event list.
Remote Extensions
If LDAP sync is enabled between the UCM6200 and the peer UCM6200,
the remote extensions will be listed under "Available Extensions". If not,
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manually enter the remote extensions under "Special Extensions" field.
Special Extensions
Manually enter the remote extensions in the peer/register trunk to be
monitored in the event list.
Valid format: 5000,5001,9000
Figure 145: Create New Event List
Remote extension monitoring works on the UCM6200 via event list BLF, among Peer SIP trunks or
Register SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the
UCM6200 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 UCM6200 and remote extensions are added to the list, the UCM6200
will send out SIP SUBSCIRBE to the remote UCM6200 to obtain the remote extension status. When the
SIP end points registers and subscribes to the local UCM6200 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 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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Note:
To configure LDAP sync, please go to UCM6200 web GUI->PBX->Basic/Call Routes->VoIP Trunk.
You will see "Sync LDAP Enable" option. Once enabled, please configure password information for the
remote peer UCM6200 to connect to the local UCM6200. Additional information such as port number,
LDAP outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM6200 and remote
UCM6200 need enable LDAP sync option with the same password for successful connection and
synchronization.
Currently LDAP sync feature only works between two UCM6200s.
(Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is
non-UCM6200 PBX. However, it might not work the other way around depending on whether the
non-UCM6200 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 get in touch with 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 GUI->PBX->Call Features->Dial By
Name.
Figure 146: Create Dial By Name Group
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1. Group Name
Enter the Group Name. This is to identify the Dial By Name group. 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 147: Dial By Name Group In IVR Key Pressing Events
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Figure 148: Dial By Name Group In Inbound Rule
2. Extension
Configure the direct dial extension for the Dial By Name group.
3. 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
GUI->PBX->Basic/Call Routes->Extensions 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 doesn't want to receive them directly.
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Figure 149: Configure Extension First Name and Last Name
4. 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.
By Full Name: enter the first 3 digits of the first name or last name to search the directory.
5. 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's the destination party, or press * to listen to the next
matching result if it's 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.
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ACTIVE CALLS AND MONITOR
The active calls on the UCM6200 are displayed in web UI->Status->Active 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 GUI->Status->Active Calls. The following figure shows
extension 1000 is calling 1001. 1001 is ringing.
Figure 150: Status->PBX Status->Active Calls - Ringing
The following figure shows the call between 1000 and 1001 is established.
Figure 151: Status->PBX Status->Active Calls – Call Established
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In active call web page, click on to refresh the active call status.
HANG UP ACTIVE CALLS
To hang up an active call, click on icon in the active call dialog. Users can also click on
to hang up all active calls.
CALL MONITOR
During an active call, click on icon and the monitor dialog will pop up.
Figure 152: 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.
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.
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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 75: UCM6200 Feature Codes] and [ENABLE SPY] section for instructions.
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CALL FEATURES
The UCM6200 supports call recording, transfer, call forward, call park and other call features via feature
code. This section lists all the feature codes in the UCM6200 and describes how to use the call features.
FEATURE CODES
Table 75: UCM6200 Feature Codes
Feature Maps
Blind Transfer
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.
Attended Transfer
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.
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.
Disconnect
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.
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Call Park
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.
Audio Mix Record
Default code: *3.
Enter the code followed by # or SEND to start recording the
audio call and the UCM6200 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.
DND/Call Forward
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.
Call Forward Busy Deactivate
Default Code: *91.
Call Forward No Answer Activate
Default Code: *92.
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
Default Code: *72.
Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.
Call Forward Unconditional
Deactivate
Default Code: *73.
Feature Misc
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Feature Code Digits Timeout
Default Setting: 1000.
Configure the maximum interval (in milliseconds) between the
digits input to activate the feature code.
Call Park
Default Extension: 700.
During an active call, initiate blind transfer and then enter this
code to park the call.
Parked Lots
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.
Use Parklot as Extension
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.
Parking Timeout (s)
Default setting: 300.
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.
Feature Codes
Voicemail Access Code
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 Voicemail
Default Code: *97.
Press *97 to access the voicemail box.
Agent Pause
Default Code: *83.
Pause the agent in all call queues.
Agent Unpause
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
Default Code: *80.
To intercom an extension, enter the code followed by the
extension number.
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Blacklist Add
Default Code: *40.
To add a number to blacklist for inbound route, dial *40 and
follow the voice prompt to enter the number.
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
Default Code: **.
To pick up a call for any extension xxxx, enter the code followed
by the extension number xxxx.
Pickup Extension
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
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.
Call Completion Request
Default Code: *11
This code is for the user who wants to use Call Completion to
complete a call.
Call Completion Cancel
Default Code: *12
This code is for the user who wants to cancel Call Completion
request.
Enable Spy
Check this box to enable spy feature codes.
Listen Spy
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. The default setting is *54.
Whisper Spy
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. The
default setting is *55.
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Barge Spy
This is the feature code to join in on the call to assist both parties.
The default setting is *56.
Enable Inbound Multiple Mode
If enabled, user can switch between different inbound route modes
with feature code. By default, this option is disabled.
Inbound Default Mode
This feature code is used to switch inbound route mode to default
mode. The default setting is *61.
Inbound Mode 1
This feature code is used to switch inbound route mode to mode 1.
The default setting is *62.
CALL RECORDING
The UCM6200 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.
1. Make sure the feature code for "Audio Mix Record" is configured and enabled.
2. After establishing the call, enter the "Audio Mix Record" feature code (by default it's *3) followed by #
or SEND to start recording.
3. To stop the recording, enter the "Audio Mix Record" feature code (by default it's *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 GUI->Status->CDR. Click on to play the recording
or click on to download the recording file.
Figure 153: Download Recording File from CDR Page
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The above recorded call's recording files are also listed under the UCM6200 web GUI->CDR->Recording
Files.
Figure 154: Download Recording File from Recording Files Page
CALL PARK
The UCM6200 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 Maps->Call 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 Misc->Call 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 THE 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.
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.
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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.
----------------------------------------------------------------------------------------------------------------------------- ---------------
Caution:
“Enable Spy” allows any user to listen to any call by feature codes. This may result in the leakage of user
privacy.
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INTERNAL OPTIONS
This section describes internal options that haven't been mentioned in previous sections yet. The settings
in this section can be applied globally to the UCM6200, including general configurations, jitter buffer, RTP
settings, ports config and STUN monitor. The options can be accessed via Web GUI->PBX->Internal
Options-> General.
INTERNAL OPTIONS/GENERAL
Table 76: 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.
Operator Extension
Specify the operator extension, which will be dialed when users press 0 to
exit voicemail application. The operator extension can also be used in
IVR option.
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.
Record Prompt
If enabled, users will hear voice prompt before recording is started or
stopped. For example, before recording, the UCM6200 will play voice
prompt "The call will be recorded". The default setting is "No".
Extension Preferences
Enforce Strong Passwords
If enabled, strong password will be enforced for the password created on
the UCM6200. The default setting is enabled.
Strong Password Rules:
1. Password for voicemail, voicemail group, outbound route, DISA, call
queue and conference requires non-repetitive and non-sequential
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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
Enable Random Password
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.
Enable Auto Email To User
If enabled, UCM6200 will send Email notification to user automatically
after editing extension settings or adding a new extension.
Disable Extension Range
If set to "Yes", users could disable the extension range
pre-configured/configured on the UCM6200. The default setting is "No".
Note:
It is recommended to keep the system assignment to avoid inappropriate
usage and unnecessary issues.
Extension Ranges
The default extension range assignment is:
User Extensions: 1000-6299
User Extensions is referring to the extensions created under web
UI->PBX->Basic/Call Routes->Extensions page.
Pick Extensions: 4000-4999
This refers to the extensions that can be manually picked from end
device when being provisioned by the UCM6200. There are two
related options in zero config page->Auto Provision 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.
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Auto Provision Extensions: 5000-6299
This sets the range for "Zero Config Extension Segment" which is the
extensions can be assigned on the UCM6200 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
Fax Extensions: 7200-8200
INTERNAL OPTIONS/JITTER BUFFER
Table 77: Internal Options/Jitter Buffer
SIP Jitter Buffer
Enable Jitter Buffer
Select to enable jitter buffer on the sending side of the SIP channel. The
default setting is "No".
Jitter Buffer Size
Configure the time (in ms) to buffer. This is the jitter buffer size used in
"Fixed" jitter buffer, or used as the initial time for "adaptive" jitter buffer.
The default setting is 100.
Max Jitter Buffer
Configure the maximum time (in ms) to buffer for "Adaptive" jitter buffer
implementation, or used as the jitter buffer size for "Fixed" jitter buffer
implementation. The default setting is 200.
Implementation
Configure the jitter buffer implementation on the sending side of a SIP
channel. The default setting is "Fixed".
Fixed
The size is always equal to the value of "Max Jitter Buffer".
Adaptive
The size is adjusted automatically and the maximum value equals to
the value of "Max Jitter Buffer".
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INTERNAL OPTIONS/RTP SETTINGS
Table 78: Internal Options/RTP Settings
RTP Start
Configure the RTP port starting number. The default setting is 10000.
RTP End
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
Configure to enable or disable RTP Checksums on RTP traffic. The
default setting is "Disable".
ICE Support
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.
STUN Server
Configure STUN server address. STUN protocol is a Client/Server and
also a Request/Response protocol. It’s 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.
Valid format:
[(hostname | IP-address) [':' port]
The default port number is 3478 if not specified.
INTERNAL OPTIONS/PAYLOAD
The UCM6200 payload type for audio codecs and video codes can be configured here.
Table 79: Internal Options/Payload
AAL2-G.726
Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword
packing). The default setting is 112.
DTMF
Configured payload type for DTMF. The default setting is 101.
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G.721 Compatible
Configure to enable/disable G.721 compatible. The default setting is Yes.
G.726
Configure the payload type for G.726 if "G.721 Compatible" is disabled.
The default setting is 111.
iLBC
Configure the payload type for iLBC. The default setting is 97.
H.264
Configure the payload type for H.264. The default setting is 99.
H.263P
Configure the payload type for H.263+. The default setting is 100 103.
VP8
Configure the payload type for VP8. The default settings is 108.
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IAX SETTINGS
The UCM6200 IAX global settings can be accessed via Web GUI->PBX->IAX Settings.
IAX SETTINGS/GENERAL
Table 80: IAX Settings/General
Bind Port
Configure the port number that the IAX2 will be allowed to listen to. The
default setting is 4569.
Bind Address
Configure the address that the IAX2 will be forced to bind to. The default
setting is 0.0.0.0, which means all addresses.
IAX1 Compatibility
Select to configure IAX1 compatibility. The default setting is "No".
No Checksums
If selected, UDP checksums will be disabled and no checksums will be
calculated/checked on systems supporting this features. The default
setting is "No".
Delay Reject
If enabled, the IAX2 will delay the rejection of calls to avoid DOS. The
default setting is "No".
ADSI
Select to enable ADSI phone compatibility. The default setting is "No".
Music On Hold Interpret
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.
Music On Hold Suggest
Specify which Music On Hold class to suggest to the bridged channel
when putting the call on hold.
Bandwidth
Configure the bandwidth for IAX settings. The default setting is "Low".
IAX SETTINGS/REGISTRATION
Table 81: 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
setting is 100.
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Auto Kill
If set to "yes", the connection will be terminated if ACK for the NEW
message is not received within 2000ms. Users could also specify number
(in milliseconds) in addition to "yes" and "no". The default setting is "yes".
Authentication Debugging
If enabled, authentication traffic in debugging will not show. The default
setting 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 almost 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
Configure ToS bit for preferred IP routing.
IAX Trunk Options
Trunk Frequency
Configure the frequency of trunk frames (in milliseconds). The default
setting is 20.
Trunk Time Stamps
If enabled, time stamps will be attached to trunk frames. The default
setting is "No".
IAX SETTINGS/STATIC DEFENSE
Table 82: IAX Settings/Static Defense
Call Token Optional
Enter a single IP address (e.g., 11.11.11.11) or a range of IP addresses
(11.11.11.11/22.22.22.22) 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 Numbers
Configure the maximum number of unvalidated calls for all IP addresses.
Call Number Limits
Configure to limit the number of calls for a give IP address of IP range.
IP or IP Range
Enter the IP address (11.11.11.11) or a range of IP addresses
(11.11.11.11/22.22.22.22) to be considered for call number limits.
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SIP SETTINGS
The UCM6200 SIP global settings can be accessed via Web GUI->PBX->SIP Settings.
SIP SETTINGS/GENERAL
Table 83: SIP Settings/General
Realm For Digest
Authentication
Configure the host name or domain name for the UCM6200. Realms
MUST be globally unique according to RFC3261. The default setting is
Grandstream.
Bind UDP Port
Configure the UDP port used for SIP. The default setting is 5060.
Bind IP Address
Configure the IP address to bind to. The default setting is 0.0.0.0, which
means binding to all addresses.
Allow Guest Calls
If enabled, the UCM6200 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 UCM6200 address to dial into
the UCM6200.
Allow Transfer
If set to "No", all transfers initiated by the endpoint in the UCM6200 will be
disabled (unless enabled in peers or users). The default setting is "Yes".
MWI From
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.
SIP SETTINGS/MISC
Table 84: 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 UCM6200 gives
up. The default setting is 0, which means the UCM6200 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.
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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
UCM6200 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".
SIP SETTINGS/SESSION TIMER
Table 85: SIP Settings/Session Timer
Session Timers
Select the session timer mode. The default setting is "Accept".
The options are:
Originate
Always request and run session timer.
Accept
Run session timer only when requested by other UA.
Refuse
Do not run session timer.
Session Expire
Configure the maximum session refresh interval (in seconds). The default
setting is 1800.
Min SE
Configure the minimum session refresh interval (in seconds). The default
setting is 90.
Session Refresher
Select the session refresher to be UAC or UAS. The default setting is
UAC.
SIP SETTINGS/TCP AND TLS
Table 86: SIP Settings/TCP and TLS
TCP Enable
Configure to allow incoming TCP connections with the UCM6200. The
default setting is "No".
TCP Bind Address
Configure the IP address for TCP server to bind to. 0.0.0.0 means binding
to all interfaces. The port number is optional. If not specified, 5060 will be
used.
TLS Enable
Configure to allow incoming TLS connections with the UCM6200. The
default setting is "No".
TLS Bind Address
Configure the IP address for TLS server to bind to. 0.0.0.0 means binding
to all interfaces. The port number is optional. If not specified, 5061 will be
used.
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Note:
The IP address must match the common name (hostname) in the
certificate. Please do not bind a TLS socket to multiple IP addresses. For
details on how to construct a certificate for SIP, please refer to the
following document:
http://tools.ietf.org/html/draft-ietf-sip-domain-certs
TLS Client Protocol
Select the TLS protocol for outbound client connections. The default
setting is TLSv1.
TLS Do Not Verify
If enabled, the TLS server's certificate won't be verified when acting as a
client. The default setting is "Yes".
TLS Self-Signed CA
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
renames as "TLS.ca" automatically.
Note:
The size of the uploaded ca file must be under 2MB.
TLS Cert
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.
TLS CA Cert
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.
TLS CA List
Display a list of files under the CA Cert directory.
SIP SETTINGS/NAT
Table 87: SIP Settings/NAT
External Host
Configure a static IP address and port (optional) used in outbound SIP
messages if the UCM6200 is behind NAT. If it is a host name, it will only
be looked up once.
Use IP address in SDP
If enabled, the SDP connection will use the IP address resolved from the
external host.
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External TCP Port
Configure the externally mapped TCP port when the UCM6200 is behind
a static NAT or PAT.
External TLS Port
Configures the externally mapped TLS port when UCM6200 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
Table 88: SIP Settings/ToS
ToS For SIP
Configure the Type of Service for SIP packets. The default setting is
None.
ToS For RTP Audio
Configure the Type of Service for RTP audio packets. The default setting
is None.
ToS For RTP Video
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/Subscription
Time
Configure the maximum duration (in seconds) of incoming registration
and subscription allowed by the UCM6200. The default setting is 3600.
Min Registration/Subscription
Time
Configure the minimum duration (in seconds) of incoming registration and
subscription allowed by the UCM6200. The default setting is 60.
Enable Relaxed DTMF
Select to enable relaxed DTMF handling. The default setting is "No".
DTMF Mode
Select DTMF mode to send DTMF. The default setting is RFC2833. If
"Info" is selected, SIP INFO message will be used. If "Inband" is selected,
64-kbit codec PCMU and PCMA are required. When "Auto" is selected,
"RFC2833" will be used if offered, otherwise "Inband" will be used. The
default setting is "RFC2833".
RTP Timeout
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 doesn't apply to calls on hold.
RTP Hold Timeout
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
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should be larger than RTP Timeout. The default setting is no timeout.
Trust Remote Party ID
Configure whether the Remote-Party-ID should be trusted. The default
setting is "No".
Send Remote Party ID
Configure whether the Remote-Party-ID should be sent or not. The
default setting is "No".
Generate In-Band Ringing
Configure whether the UCM6200 should generate inband ringing or not.
The default setting is "Never".
Yes: The UCM6200 will send 180 Ringing followed by 183 Session
Progress and in-band audio.
No: The UCM6200 will send 180 Ringing if 183 Session Progress has
not been sent yet. If audio path is established already with 183 then
send in-band ringing.
Never: Whenever ringing occurs, the UCM6200 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 UCM6200.
Send Compact SIP Headers
If enabled, compact SIP headers will be sent. The default setting is "No".
100rel
Configure the 100rel setting on UCM6200. The default setting is “Yes”.
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PORTS CONFIG
The analog hardware (FXS port and FXO port) on the UCM6200 will be listed in this page. Click on to
edit 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.
Figure 155: 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" for the UCM6200 to automatically detect the ACIM value. The
detecting value will be automatically filled into the settings.
Figure 156: FXO Ports ACIM Settings
Table 89: Internal Options/Ports Config
Tone Region
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)".
Advanced Settings
FXO Opermode
Select country to set the On Hook Speed, Ringer Impedance, Ringer
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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)".
FXS Opermode
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)".
FXS TISS Override
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 600Ω.
PCMA Override
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.
Boost Ringer
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".
Fast Ringer
Configure to increase the ringing speed to 25HZ. This option can be used
with "Low Power" option. The default setting is "Normal".
Low Power
Configure the peak voltage up to 50V during "Fast Ringer" operation. This
option is used with "Fast Ringer". The default setting is "Normal".
Ring Detect
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".
FXS MWI Mode
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.
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VALUE-ADDED FEATURES
FAX SENDING
The UCM6200 supports sending Fax via web UI access. This feature can be found on web
UI->PBX->Value-added Features->Fax Sending page. In order 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 UCM6200 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 UI. Users can also view the
sending history is in the same web page.
Figure 157: Fax Sending in Web UI
ANNOUNCEMENTS CENTER
The UCM6200 supports Announcements Center feature which allows users to pre-record and store voice
message into UCM6200 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.
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Figure 158: Announcements Center
ANNOUNCEMENTS CENTER SETTINGS
Table 90: Announcements Center Settings
Name
Configure a name for the newly created Announcements Center to
identify this announcement center.
Code
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.
Custom Prompt
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.
Ring Timeout
Configure the ring timeout for the group members. The default value is 30
seconds.
GROUP SETTINGS
Table 91: Group Settings
Name
Configure a name for the newly created group to identify the group.
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Number
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.
Announcements Center feature can be found under web UI->PBX->Value-added Features->
Announcements Center. The following example demonstrates the usage of this feature.
1. Click to create new group.
2. Give a name to the newly created group.
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 159: Announcements Center Group Configuration
In this example, group “Test” has number 666. Extension 1000, 1001 and 1002 are in this group.
5. Click to create a new Announcement Center.
6. Give a name to the newly created Announcement Center.
7. Specify the code which will be used with group number to send the voice message to.
8. Select the message that will be used by the code from the Custom Prompt drop down menu. To create
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a new Prompt, please click “Prompt” link and follow the instructions in that page.
Figure 160: 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 UCM6200. 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 161: Announcements Center Example
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STATUS AND REPORTING
PBX STATUS
The UCM6200 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
GUI->Status->PBX Status.
Figure 162: Status->PBX Status
TRUNKS
Users could see all the configured trunk status in this section.
Figure 163: Trunk Status
Table 92: Trunk Status
Status
Display trunk status.
Analog trunk status:
Available
Busy
Unavailable
Unknown Error
SIP Peer trunk status:
Unreachable: The hostname cannot be reached.
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Unmonitored: QUALIFY feature is not turned on to be monitored.
Reachable: The hostname can be reached.
SIP Register trunk status:
Registered
Unrecognized Trunk
Trunks
Display trunk name
Type
Display trunk Type:
Analog
SIP
IAX
Username
Display username for this trunk.
Port/Hostname/IP
Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.
Other operations are also available in trunk status section:
Click on "Trunks", the web page will redirect to trunk configuration page which can also be accessed
via web GUI->PBX->Basic/Call Routes->Analog Trunks.
Click on to refresh the trunk status.
Click on [ + ] to expand the status detail table.
Click on [ - ] to hide the status detail table.
EXTENSIONS
Users could see all the configured extension status in this section.
Figure 164: Extension Status
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Table 93: Extension Status
Status
Display extension number (including feature code). The color indicator has the
following definitions.
Green: Free
Blue: Ringing
Yellow: In Use
Grey: Unavailable
Extension
Display the extension number.
Name/Label
First name and last name of the extension.
Message
Display message status for the extension.
Example: 2/4/1
Description: There are 2 urgent messages, 4 messages in total and 1
message that has been already read.
Type
Displays extension type.
SIP User
IAX User
Analog User
Ring Groups
Voicemail Groups
Other operations are also available in extension status section:
Click on "Extensions", the web page will redirect to extension configuration page which can also be
accessed via web GUI->PBX->Basic/Call Routes->Extensions.
Click on to refresh the extension status.
Click on one of the tabs to display
the corresponding extensions accordingly.
Click on [ + ] to expand the status detail table.
Click on [ - ] to hide the status detail table.
QUEUES
Users could see all the configured call queue status in this section. The following figure shows the call
queue 6500 being in used.
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Figure 165: Queue Status
The current call status (caller ID, duration), agent status, service level, calls summary
(completed/abandoned) are shown for the call queue. The agent status is defined as below.
Table 94: Agent Status
The agent is available/idle.
The agent is ringing.
The agent is talking/busy.
The agent has been logged out.
On the UCM6200, Service Level is defined as the percentage of high-quality calls over all calls in the call
queue, where high-quality call means calls answered within 10 seconds.
Other operations are also available in queue status section:
Click on "Queues", the web page will redirect to call queue configuration page which can also be
accessed via web GUI->PBX->Call Features->Call Queue.
Click on to refresh the call queue status.
Click on [ + ] to expand the call queue detail.
Click on [ - ] to hide the call queue detail.
CONFERENCE ROOMS
Users could see all the conference room status in this section. It shows all the configured conference
rooms, current users, call duration for each user and conference call.
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Figure 166: Conference Room Status
Other operations are also available in conference room status section:
Click on "Conference Rooms", the web page will redirect to conference room configuration page which
can also be accessed via web GUI->PBX->Call Features->Conference.
Click on to refresh the conference room status.
Click on [ + ] to expand the conference room details.
Click on [ - ] to hide the conference room details.
INTERFACES STATUS
This section displays interface/port connection status on the UCM6200. The following example shows the
interface status for UCM6204 with USB, WAN port, FXS1, FXS2 and FXO1 connected.
Figure 167: UCM6204 Interfaces Status
Table 95: Interface Status Indicators
USB connected.
USB disconnected.
SD Card connected.
SD Card disconnected.
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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.
Other operations are also available in interface status section:
Click on "Interfaces Status", the web page will redirect to ports configuration page which can also be
accessed via web GUI->PBX->Internal Options->Ports Config.
Click on to refresh the interface status.
Click on [ + ] to expand the interface details.
Click on [ - ] to hide the interface details.
PARKING LOT
The UCM6200 supports call park using feature code. When there is call being parked, this section will
display the parking lot status.
Figure 168: Parking Lot Status
Table 96: Parking Lot Status
Caller ID
Display the caller ID who parks the call.
Channel
Display channel for the call park.
Extension
Display the parking lot number where the call is parked/retrieved.
Timeout
Display timeout (in seconds) for the parked call. The status page will
dynamically update this timer from 120 seconds (default) to 0. When the timer
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reaches 0, the caller who parks the call will be called back.
Other operations are also available in parking lot status section:
Click on "Parking Lot", the web page will redirect to feature codes page which can also be accessed
via web GUI->PBX->Internal Options->Feature Codes.
Click on to refresh the parking lot status.
Click on [ + ] to expand the parking lot details.
Click on [ - ] to hide the parking details.
SYSTEM STATUS
The UCM6200 system status can be accessed via Web GUI->Status->System Status, which displays the
following system information.
General
Network
Storage Usage
Resource Usage
GENERAL
Under Web GUI->Status->System Status->General, users could check the hardware and software
information for the UCM6200. Please see details in the following table.
Table 97: System Status->General
Status ->System Status -> General
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.
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Recovery
Recovery version.
NETWORK
Under Web GUI->Status->System Status->Network, users could check the network information for the
UCM6200. Please see details in the following table.
Table 98: System Status->Network
Status -> System Status -> Network
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.
IP Address
IP address.
Gateway
Default gateway address.
Subnet Mask
Subnet mask address.
DNS Server
DNS Server address.
STORAGE USAGE
Users could access the storage usage information from web UI->Status->System Status->Storage
Usage. It shows the available and used space for the following partitions.
Configuration partition
This partition contains PBX system configuration files and service configuration files.
Data partition
Voicemail, recording files, IVR file, Music on Hold files and etc.
USB disk
USB disk will display if connected.
SD Card
SD Card will display if connected.
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Figure 169: System Status->Storage Usage
RESOURCE USAGE
When configuring and managing the UCM6200, users could access resource usage information to
estimate the current usage and allocate the resources accordingly. Under web UI->Status->System
Status->Resource Usage, the current CPU usage and Memory usage are shown in the pie chart.
Figure 170: System Status->Resource Usage
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SYSTEM EVENTS
The UCM6200 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->Status->System Events->Alert Events List.
The following event are currently supported on the UCM6200 which will have alert and/or Email generated
if occurred:
Disk Usage
Modify Admin Password
Memory Usage
System Reboot
System Update
System Crash
Register SIP Failed
Register SIP Trunk Failed
Restore Config
User Login Success
User Login Failed
SIP Internal Call Failure
SIP Outgoing Call through Trunk Failure
Fail2ban Blocking
SIP Lost Registration
SIP Peer Trunk Status
Click on to configure the parameters for each event. See examples below.
1. Disk Usage.
Figure 171: System Events->Alert Events Lists: Disk Usage
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Detect Cycle: The UCM6200 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 UCM6200
system will send the alert.
2. Memory Usage
Figure 172: System Events->Alert Events Lists: Memory Usage
Detect Cycle: The UCM6200 will perform the memory 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 UCM6200
system will send the alert.
3. System Reboot
Figure 173: System Events->Alert Events Lists: System Reboot
Detect Cycle: The UCM6200 will check the system reboot based on this cycle. Users can enter
the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
4. System Crash
Figure 174: System Events->Alert Events Lists: System Crash
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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.
ALERT LOG
Under Web GUI->Status->System Events->Alert Log, system messages from triggered system events
are listed as alert logs. The following screenshot shows system crash alert logs.
Figure 175: System Events->Alert 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 . Alert logs are
classified into two types by the system:
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”.
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Figure 176: Filter for Alert Log
ALERT CONTACT
Users could add administrator's Email address under Web GUI->Status->System Events->Alert Contact
to send the alert notification to. Up to 10 Email addresses can be added.
CDR
A Call Detail Record (CDR) is a data record produced by telephone exchange activities or other
telecommunications equipment documenting the details of a phone call that passed through the PBX. The
CDR is composed of the following data fields on the UCM6200.
Start Time. Format: 2016-02-27 16:47:03.
Call From. Format: "John Doe"<6012>.
Call To. Format: 6005.
Answered By. Format: 6005.
Call Time. Format: 0:00:10.
Talk Time. Format: 0:00:10
Status. Format: NO ANSWER, BUSY, ANSWERED, or FAILED.
Options. Voice record playing/downloading/deleting.
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. Then click on "View Report" button to display the generated
report.
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Figure 177: CDR Filter
Table 99: CDR Filter Criteria
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.
Inbound Trunks
Select certain inbound trunk(s) and the CDR of calls going inbound through the
trunk(s) will be filtered out.
Outbound Trunks
Select certain outbound trunk(s) and the CDR of calls going outbound through the
trunk(s) will be filtered out.
Start Time
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.
End 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.
Caller Number
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,
case-insensitive, 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.
Caller Name
Enter the caller name to filter the CDR report. CDR with the matching caller name will
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be filtered out.
Callee Number
Enter the callee number to filter the CDR report. CDR with the matching callee
number will be filtered out.
The call report will display as the following figure shows.
Figure 178: Call Report
Users could perform the following operations on the call report.
Sort
Click on the header of the column to sort by this category. For example, clicking on "Start Time" will
sort the report according to start time. Clicking on "Start Time" again will 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
On the bottom of the page, click on "Delete All" button to remove all the call report information.
Play/Download/Delete Recording File (per entry)
If the entry has audio recording file for the call, the three icons on the most right 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 179: Call Report Entry with Audio Recording File
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Automatic Download CDR Records
User could configure the UCM6200 to automatically download the CDR records and send the records
to an Email address. Click on “Automatic Download Settings”, and configure the parameters in the
dialog below.
Figure 180: Automatic Download Settings
To receive CDR record automatically from Email, check “Enable” and select a time period “By Day” “By
Week” or “By Month” for the automatic download period. Make sure you have entered an Email
address to receive the CDR records.
CDR IMPROVEMENT
Starting from UCM6200 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
UI-> Status->CDR->CDR. 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 181: CDR Report
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Figure 182: Detailed CDR Information
DOWNLOADED CDR FILE
The downloaded CDR (.csv file) has different format from the web UI CDR. Here are some descriptions.
Call From, Call To
"Call From": the caller ID.
"Call To": the callee ID.
If "Call From" shows empty, "Call To" shows "s" (see highlight part in the picture below) and the "Source
Channel" contains "DAHDI", this means the call is from FXO/PSTN line. For FXO/PSTN line, we only know
there is an incoming request when there is incoming call but we don't know the number being called. So
we are using "s" to match it where "s" means "start".
Figure 183: Downloaded CDR File Sample - Call To Shows "s"
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:
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Figure 184: Downloaded CDR File Sample - Source Channel and Dest Channel 1
DAHDI means it is an analog call, FXO or FXS.
For UCM6202, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports.
For UCM6204, DAHDI/(1-4) are FXO ports, and DAHDI(5-6) are FXS ports.
For UCM6208, DAHDI/(1-8) are FXO ports, and DAHDI(9-10) are FXS ports.
Sample 2:
Figure 185: Downloaded CDR File Sample - Source Channel and Dest Channel 2
"SIP" means it's a SIP call. There are three possible format:
(a) SIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string
and can be ignored.
(c) SIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out
through the trunk.
(c) SIP/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.
Sample 3:
Figure 186: Downloaded CDR File Sample - Source Channel and Dest Channel 3
This is a very special channel name. If it shows up, most likely it means a conference call.
There are some other possible values, but these values are almost 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
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STATISTICS
CDR Statistics is an additional feature on the UCM6200 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.
Figure 187: CDR Statistics
Table 100: CDR Statistics Filter Criteria
Trunk Type
Select one of the following trunk type.
All
SIP Calls
PSTN Calls
Call Type
Select one or more in the following checkboxes.
Inbound calls
Outbound calls
Internal calls
External calls
All calls
Time Range
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.
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
UCM6200.
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Figure 188: CDR->Recording 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 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.
API CONFIGURATION
The UCM6200 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.
Before accessing the API, the administrators need enable API and configure the access/authentication
information on the UCM6200 first. The API configuration parameters are listed in the table below.
Table 101: API Configuration Files
Enable
Enable/Disable API. The default setting is disabled.
TLS Bind Address
Configure the IP 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 8443. The IP
address must match the common name (host name) in the certificate so that the TLS
socket won't bind to multiple IP addresses. The default setting is 0.0.0.0:8443.
TLS Private Key
Upload TLS private key. The size of the key file must be under 2MB. This file will be
renamed as 'private.pem' automatically.
TLS Cert
Upload TLS cert. The size of the certificate must be under 2MB. This is the certificate
file (*.pem format only) for TLS connection. This file will be renamed as
"certificate.pem" automatically. It contains private key for the client and signed
certificate for the server.
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Username
Configure the Username for API Authentication.
Password
Configure the Password for API Authentication.
Permitted
Specify a list of IP addresses permitted by API. This creates an AIP-specific access
control list. Multiple entries are allowed.
For example, "192.168.40.3/255.255.255.255" denies access from all IP addresses
except 192.168.40.3.
The default setting is blank, meaning all IPs will be denied. Users must set permitted
IP address before connecting to the API.
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/ucm6200_cdr_rec_api_guide.pdf
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UPGRADING AND MAINTENANCE
UPGRADING
The UCM6200 can be upgraded to a new firmware version remotely or locally. This section describes how
to upgrade your UCM6200 via network or local upload.
UPGRADING VIA NETWORK
The UCM6200 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the
TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP, HTTP or
HTTPS; the server name can be FQDN or IP address.
Examples of valid URLs:
firmware.grandstream.com/BETA
The upgrading configuration can be accessed via Web GUI->Maintenance->Upgrade.
Figure 189: Network Upgrade
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Table 102: Network Upgrade Configuration
Upgrade Via
Allow users to choose the firmware upgrade method: TFTP, HTTP or
HTTPS.
Firmware Server Path
Define the server path for the firmware server.
Firmware File Prefix
If configured, only the firmware with the matching encrypted prefix will be
downloaded and flashed into the UCM6200.
Firmware File Suffix
If configured, only the firmware with the matching encrypted postfix will be
downloaded and flashed into the UCM6200.
HTTP/HTTPS User Name
The user name for the HTTP/HTTPS server.
HTTP/HTTPS Password
The password for the HTTP/HTTPS server.
Please follow the steps below to upgrade the firmware remotely.
Enter the firmware server path under web UI->Maintenance->Upgrade.
Click on "Save". Then reboot the device to start the upgrading process.
Please be patient during the upgrading process. Once done, a reboot message will be displayed in the
LCD.
Manually reboot the UCM6200 when it's appropriate to avoid immediate service interruption. After it
boots up, log in the web GUI to check the firmware version.
UPGRADING VIA LOCAL UPLOAD
If there is no HTTP/TFTP server, users could also upload the firmware to the UCM6200 directly via Web
GUI. Please follow the steps below to upload firmware locally.
Download the latest UCM6200 firmware file from the following link and save it in your PC.
http://www.grandstream.com/support/firmware
Log in the Web GUI as administrator in the PC.
Go to Web GUI->Maintenance->Upgrade, upload the firmware file by clicking on and select
the firmware file from your PC. The default firmware file name is ucm6200fw.bin
Figure 190: Local Upgrade
Click on to start upgrading.
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Figure 191: Upgrading Firmware Files
Wait until the upgrading process is successful and a window will be popped up in the Web GUI.
Figure 192: Reboot UCM6200
Click on "OK" to reboot the UCM6200 and check the firmware version after it boots up.
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Note:
Please do not interrupt or power cycle the UCM6200 during upgrading process.
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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 UCM6200 to the same LAN segment;
3. Launch the TFTP server and go to the File menu->Configure->Security 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 UCM6200 web configuration interface;
5. Configure the Firmware Server Path to the IP address of the PC;
6. Update the changes and reboot the UCM6200.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use
Microsoft IIS web server.
BACKUP
The UCM6200 configuration can be backed up locally or via network. The backup file will be used to
restore the configuration on UCM6200 when necessary.
BACKUP/RESTORE
Users could backup the UCM6200 configurations for restore purpose under Web
GUI->Maintenance->Backup->Local Backup.
Click on to create a new backup file. Then the following dialog will show.
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Figure 193: 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 or Local.
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 UCM6200 internal storage or the external
device.
Click on to upload backup file from the local device to UCM6200. The uploaded
backup file will also be displayed in the web page and can be used to restore the UCM6200.
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Figure 194: Backup / Restore
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.
Figure 195: Local Backup
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DATA SYNC
Besides local backup, users could backup the voice records/voice mails/CDR/FAX in a daily basis to a
remote server via SFTP protocol automatically under Web GUI->Maintenance->Backup->Data Sync.
Figure 196: Data Sync
Table 103: Data Sync Configuration
Enable Data Sync
Enable the auto data sync function. The default setting is "No".
Account
Enter the Account name on the SFTP backup server.
Password
Enter the Password associate with the Account on the SFTP backup
server.
Server Address
Enter the SFTP server address.
Sync Time
Enter 0-23 to specify the backup hour of the day.
Before saving the configuration, users could click on "Test Connection". The UCM6200 will then try
connecting the server to make sure the server is up and accessible for the UCM6200. Save the changes
and all the backup logs will be listed on the web page. After data sync is configured, users could also
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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 UCM6200 from a backup file, users could go to Web
GUI->Maintenance->Backup->Local Backup.
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 UCM6200.
If users have other backup files on PC to restore on the UCM6200, click on "Upload Backup File" first
and select it from local PC to upload on the UCM6200. Once the uploading is done, this backup file will
be displayed in the list of previous configuration backups for restore purpose. Click on to restore
from the backup file.
Figure 197: Restore UCM6200 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.
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CLEANER
Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under
Web GUI->Maintenance->Cleaner.
Figure 198: Cleaner
Table 104: Cleaner Configuration
Enable CDR Cleaner
Enable the CDR Cleaner function.
CDR Clean Time
Enter 0-23 to specify the hour of the day to clean up CDR.
Clean Interval
Enter 1-30 to specify the day of the month to clean up CDR.
Enable VR Cleaner
Enter the Voice Records Cleaner function.
Choose Cleaner File
Select the files for system automatic clean.
Recording Files
Conference
Queue
Voicemail
Fax
VR Clean Threshold
Specify the Voice Records threshold from 0 to 99 by using local storage
status in percentage.
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VR Clean Time
Enter 0-23 to specify the hour of the day to clean up Voice Records.
Clean Interval
Enter 1-30 to specify the day of the month to clean up Voice Records.
All the cleaner logs will be listed on the bottom of the page.
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Note:
Cleaner will delete data based on Recording Storage selection. If USB Disk is selected, Cleaner will only
clean data in USB and local data will leave untouched. If Enable auto change is selected and USB disk is
connected, Cleaner will only delete data in USB drive. Recordings Storage function can be found under
web UI-> Settings-> Recordings Storage-> Recordings Storage.
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RESET AND REBOOT
Users could perform reset and reboot under Web GUI->Maintenance->Reset and Reboot.
To factory reset the device, select the mode type first. There are two different types for reset.
User Data: All the data including voicemail, recordings, IVR Prompt, Music on Hold, CDR and backup
files will be cleared.
All: All the configurations and data will be reset to factory default.
Figure 199: Reset and Reboot
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SYSLOG
On the UCM6200, users could dump the syslog information to a remote server under Web
GUI->Maintenance->Syslog. Enter the syslog server hostname or IP address and select the module/level
for the syslog information.
The default syslog level for all modules is "error", which is recommended in your UCM6200 settings
because it can be helpful to locate the issues when errors happen.
Some typical modules for UCM6200 functions are as follows and users can turn on "notic" and "verb"
levels besides "error" level.
pbx: This module is related to general PBX functions.
chan_sip: This module is related to SIP calls.
chan_dahdi: This module is related to analog calls (FXO/FXS).
app_meetme: This module is related to conference bridge.
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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 print might cause traffic and affect system
performance.
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TROUBLESHOOTING
On the UCM6200, users could capture traces, ping remote host and traceroute remote host for
troubleshooting purpose under Web GUI->Maintenance->Troubleshooting.
ETHERNET CAPTURE
The captured trace can be downloaded for analysis. Also the instructions or result will be displayed in the
web GUI output result.
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Figure 200: Ethernet Capture
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 capturing the
trace.
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 201: 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 202: Traceroute
ANALOG RECORD TRACE
Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM6200 user has
caller ID issue for incoming call from Analog trunk. Users can access analog record trance under web
GUI->Maintenance->Troubleshooting ->Analog Record Trace.
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. Select "Record Direction".
3. Select "Record File Mode" to separate the record per direction or mix.
4. Click on "Start".
5. Make a call via the analog port that has the issue.
6. Once done, click on "Stop".
7. Click on "Download" to download the analog record trace.
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Figure 203: Troubleshooting Analog Trunks
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 UCM6200. 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
UCM6200. The default setting is 3. If there is no response from UCM6200 after 3 attempts (default) to
check, current status will be stored and the internal service in UCM6200 will be restarted.
Figure 204: Service Check
NETWORK STATUS
In UCM6200 web UI->Maintenance->Troubleshooting->Network Status, the users can view active Internet
connections. This information can be used to troubleshoot connection issue between UCM6200 and other
services.
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Figure 205: Network Status
REMOTE ACCESS
SSH ACCESS
SSH switch now is available via web UI and LCD. User can enable or disable SSH access directly from
web UI or LCD screen. For web SSH access, please log in UCM6200 web interface and go to
Maintenance->Remote Access->SSH Access. By default, SSH access is disabled for security concerns.
It is highly recommended to only enable SSH access for debugging purpose.
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Figure 206: SSH Access
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EXPERIENCING THE UCM6200 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 UCM6200 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.
FCC Caution:
This device complies with part15 of the FCC Rules. Operation is subject to the following two
conditions: (1) This device may not cause harmful interference, and (2) this device must accept any
interference received, including interference that may cause undesired operation.
This equipment has been tested and found to comply with the limits for a Class B digital device,
pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection
against harmful interference in a residential installation. This equipment generates, uses and can
radiate radio frequency energy and, if not installed and used in accordance with the instructions, may
cause harmful interference to radio communications. However, there is no guarantee that
interference will not occur in a particular installation. If this equipment does cause harmful
interference to radio or television reception, which can be determined by turning the equipment off
and on, the user is encouraged to try to correct the interference by one or more of the following
measures:
—Reorient or relocate the receiving antenna.
—Increase the separation between the equipment and receiver.
—Connect the equipment into an outlet on a circuit different from that to which the receiver is
connected.
—Consult the dealer or an experienced radio/TV technician for help.
Any changes or modifications not expressly approved by the party responsible for compliance could
void the user's authority to operate the equipment.
Regulatory Information
U.S. FCC Part 68 Statement
This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. The
unit bears a label on the back which contains among other information a product identifier in the format
US: GNIIS00BUCM6202. If requested, this number must be provided to the telephone company.
This equipment uses the following standard jack types for network connection: RJ11C.
This equipment contains an FCC compliant modular jack. It is designed to be connected to the telephone
network or premises wiring using compatible modular plugs and cabling which comply with the
requirements of FCC Part 68 rules.
The Ringer Equivalence Number, or REN, is used to determine the number of devices which may be
connected to the telephone line. An excessive REN may cause the equipment to not ring in response to
an incoming call. In most areas, the sum of the RENs of all equipment on a line should not exceed five
(5.0).
In the unlikely event that this equipment causes harm to the telephone network, the telephone company
can temporarily disconnect your service. The telephone company will try to warn you in advance of any
such disconnection, but if advance notice isn't practical, it may disconnect the service first and notify you
as soon as possible afterwards. In the event such a disconnection is deemed necessary, you will be
advised of your right to file a complaint with the FCC.
From time to time, the telephone company may make changes in its facilities, equipment, or operations
which could affect the operation of this equipment. If this occurs, the telephone company is required to
provide you with advance notice so you can make the modifications necessary to obtain uninterrupted
service.
There are no user serviceable components within this equipment. See Warranty flyer for repair or
warranty information.
It shall be unlawful for any person within the United States to use a computer or other electronic device
to send any message via a telephone facsimile unless such message clearly contains, in a margin at the
top or bottom of each transmitted page or on the first page of the transmission, the date and time it is sent
and an identification of the business, other entity, or individual sending the message and the telephone
number of the sending machine or of such business, other entity, or individual. The telephone number
provided may not be a 900 number or any other number for which charges exceed local or long distance
transmission charges. Telephone facsimile machines manufactured on and after December 20, 1992,
must clearly mark such identifying information on each transmitted message. Facsimile modem boards
manufactured on and after December 13, 1995, must comply with the requirements of this section.
This equipment cannot be used on public coin phone service provided by the telephone company.
Connection to Party Line Service is subject to state tariffs. Contact your state public utility commission,
public service commission, or corporation commission for more information.