972-770-4600
Grandstream Networks, Inc.
UCM6200 Series IP PBX
User Manual
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COPYRIGHT
©2017 Grandstream Networks, Inc. http://www.grandstream.com
All rights reserved. Information in this document is subject to change without notice. Reproduction or transmittal
of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express
written permission of Grandstream Networks, Inc. is not permitted.
The latest electronic version of this user manual is available for download here:
http://www.grandstream.com/support
Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc. in
the United States, Europe and other countries.
CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this product
in any way other than as detailed by this User Manual, could void your manufacturer warranty.
WARNING
Please do not use a different power adaptor with your devices as it may cause damage to the products and
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void the manufacturer warranty.
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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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Table of Content
DOCUMENT PURPOSE........................................................................... 22
CHANGE LOG ......................................................................................... 23
Firmware Version 1.0.15.14................................................................................................................ 23
Firmware Version 1.0.15.13................................................................................................................ 23
Firmware Version 1.0.14.23................................................................................................................ 24
Firmware Version 1.0.14.21................................................................................................................ 24
Firmware Version 1.0.13.14................................................................................................................ 24
Firmware Version 1.0.12.19................................................................................................................ 25
Firmware Version 1.0.11.27 ................................................................................................................ 26
Firmware Version 1.0.0.7 .................................................................................................................... 26
WELCOME ............................................................................................... 27
PRODUCT OVERVIEW ............................................................................ 28
Technical Specifications ..................................................................................................................... 28
INSTALLATION ........................................................................................ 31
Equipment Packaging......................................................................................................................... 31
Connect Your UCM6200 ..................................................................................................................... 31
Connect The UCM6202 .............................................................................................................. 31
Connect The UCM6204 .............................................................................................................. 32
Connect The UCM6208 .............................................................................................................. 33
Safety Compliances ............................................................................................................................ 34
Warranty ............................................................................................................................................. 34
GETTING STARTED ................................................................................ 35
Use the LCD Menu ............................................................................................................................. 35
Use the LED Indicators ....................................................................................................................... 37
Use the Web GUI ............................................................................................................................... 38
Access Web GUI ......................................................................................................................... 38
Setup Wizard............................................................................................................................... 40
Web GUI Configurations ............................................................................................................. 40
Web GUI Languages .................................................................................................................. 41
Save and Apply Changes............................................................................................................ 41
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Make Your First Call ........................................................................................................................... 42
SYSTEM SETTINGS ................................................................................ 43
Network Settings ................................................................................................................................ 43
Basic Settings ............................................................................................................................. 43
DHCP Client List ......................................................................................................................... 48
802.1X ......................................................................................................................................... 49
Static Routes ............................................................................................................................... 51
Port Forwarding .......................................................................................................................... 53
OpenVPN .................................................................................................................................... 55
DDNS Settings ............................................................................................................................ 57
Security Settings ................................................................................................................................. 59
Static Defense ............................................................................................................................. 59
Dynamic Defense ........................................................................................................................ 62
Fail2ban ...................................................................................................................................... 63
SSH Access ................................................................................................................................ 65
LDAP Server ....................................................................................................................................... 66
LDAP Server Configurations ....................................................................................................... 66
LDAP Phonebook ........................................................................................................................ 68
LDAP Client Configurations ........................................................................................................ 71
HTTP Server ....................................................................................................................................... 74
Time settings ...................................................................................................................................... 74
Auto time updating ...................................................................................................................... 74
Set Time Manually ...................................................................................................................... 76
NTP Server ................................................................................................................................. 76
Office Time .................................................................................................................................. 76
Holiday ........................................................................................................................................ 78
Email ................................................................................................................................................... 80
Email settings .............................................................................................................................. 80
Email Templates .......................................................................................................................... 81
Email Send Log ........................................................................................................................... 83
Recordings Storage ............................................................................................................................ 84
Google Service Settings Support ....................................................................................................... 86
PROVISIONING ....................................................................................... 89
Overview ............................................................................................................................................. 89
Configuration Architecture for End Point Device ................................................................................ 89
Auto Provisioning Settings.................................................................................................................. 90
Discovery ............................................................................................................................................ 93
Uploading Devices List ....................................................................................................................... 95
Managing discovered devices: ........................................................................................................... 95
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Global configuration ............................................................................................................................ 96
Global policy................................................................................................................................ 96
Global Templates ...................................................................................................................... 103
Model configuration .......................................................................................................................... 105
Model templates ........................................................................................................................ 105
Model Update ............................................................................................................................ 107
Device Configuration ........................................................................................................................ 109
Create New Device ................................................................................................................... 109
Manage Devices ....................................................................................................................... 110
Sample Application ........................................................................................................................... 116
EXTENSIONS......................................................................................... 121
Create New User .............................................................................................................................. 121
Create New SIP Extension ....................................................................................................... 121
Create New IAX Extension ....................................................................................................... 129
Create New FXS Extension ...................................................................................................... 133
Batch Add Extensions ....................................................................................................................... 138
Batch Add SIP Extensions ........................................................................................................ 138
Batch Add IAX Extensions ........................................................................................................ 141
Search and Edit Extension ............................................................................................................... 143
Export Extensions ............................................................................................................................. 144
Import Extensions ............................................................................................................................. 145
E-mail Notification ............................................................................................................................. 152
Multiple Registrations per Extension ................................................................................................ 153
SMS Message Support ..................................................................................................................... 155
EXTENSION GROUPS........................................................................... 156
Configure Extension Groups ............................................................................................................ 156
Using Extension Groups ................................................................................................................... 157
ANALOG TRUNKS ................................................................................ 158
Analog Trunk Configuration .............................................................................................................. 158
PSTN Detection ................................................................................................................................ 161
VOIP TRUNKS ....................................................................................... 164
VoIP Trunk Configuration .................................................................................................................. 164
Direct Outward Dialing (DOD) .......................................................................................................... 174
SLA STATION ........................................................................................ 176
Create/Edit SLA Station .................................................................................................................... 176
Sample Configuration ....................................................................................................................... 177
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CALL ROUTES ...................................................................................... 179
Outbound Routes ............................................................................................................................. 179
Configuring Outbound Routes .................................................................................................. 179
Outbound Blacklist .................................................................................................................... 182
PIN Groups ............................................................................................................................... 182
Inbound Routes ................................................................................................................................ 187
Inbound Rule Configurations .................................................................................................... 187
Inbound Route: Prepend Example ............................................................................................ 189
Inbound Route: Multiple Mode .................................................................................................. 190
FAX Intelligent Route ................................................................................................................ 192
FAX with Two Media.................................................................................................................. 192
Blacklist Configurations ............................................................................................................. 192
CONFERENCE ROOM ........................................................................... 194
Conference Room Configurations............................................................................................. 194
Join a Conference Call .............................................................................................................. 196
Invite Other Parties to Join Conference .................................................................................... 197
During The Conference ............................................................................................................. 198
Record Conference ................................................................................................................... 199
CONFERENCE SCHEDULE .................................................................. 200
Conference Schedule Configuration ................................................................................................ 200
IVR ......................................................................................................... 204
Configure IVR ................................................................................................................................... 204
Black/White List in IVR ..................................................................................................................... 207
Create Custom Prompt ..................................................................................................................... 209
LANGUAGE SETTINGS FOR VOICE PROMPT .................................... 210
Download and Install Voice Prompt Package................................................................................... 210
Customize Specific Prompt .............................................................................................................. 212
VOICEMAIL ............................................................................................ 213
Configure Voicemail .......................................................................................................................... 213
Access Voicemail .............................................................................................................................. 215
Extension Voicemail Count ............................................................................................................... 216
Voicemail Email Settings .................................................................................................................. 216
Configure Voicemail Group............................................................................................................... 217
RING GROUP......................................................................................... 219
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Configure Ring Group ....................................................................................................................... 219
Remote Extension in Ring Group ..................................................................................................... 221
PAGING AND INTERCOM GROUP ....................................................... 224
Configure Paging/Intercom Group .................................................................................................... 224
CALL QUEUE ........................................................................................ 226
Configure Call Queue ....................................................................................................................... 226
Call Center Settings and enhancements .......................................................................................... 230
Queue Statistics ............................................................................................................................... 232
Switchboard ...................................................................................................................................... 233
PICKUP GROUPS .................................................................................. 235
Configure Pickup Groups ................................................................................................................. 235
Configure Pickup Feature Code ....................................................................................................... 235
MUSIC ON HOLD ................................................................................... 237
FAX SERVER ......................................................................................... 240
Configure Fax/T.38 ........................................................................................................................... 240
Receiving Fax ................................................................................................................................... 242
Sample Configuration to Receive Fax from PSTN Line ........................................................... 242
Sample Configuration for Fax-To-Email .................................................................................... 244
FAX Sending ..................................................................................................................................... 246
BUSY CAMP-ON .................................................................................... 247
PRESENCE ............................................................................................ 248
FOLLOW ME .......................................................................................... 250
SPEED DIAL .......................................................................................... 252
DISA ....................................................................................................... 253
CALLBACK ............................................................................................ 255
BLF AND EVENT LIST........................................................................... 256
BLF ................................................................................................................................................... 256
Event List .......................................................................................................................................... 256
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DIAL BY NAME ...................................................................................... 259
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Dial by Name Configuration.............................................................................................................. 259
User Name Prompt Customization ................................................................................................... 262
Upload User Name Prompt File from Web GUI ........................................................................ 262
Record User Name via Voicemail Menu ................................................................................... 262
ACTIVE CALLS AND MONITOR ........................................................... 263
Active Calls Status ............................................................................................................................ 263
Hang Up Active Calls ........................................................................................................................ 264
Call Monitor ...................................................................................................................................... 264
CALL FEATURES .................................................................................. 266
Feature Codes .................................................................................................................................. 266
Call Recording .................................................................................................................................. 271
Call Park ........................................................................................................................................... 272
Park a Call................................................................................................................................. 272
Retrieve Parked Call ................................................................................................................. 272
Enable Spy ....................................................................................................................................... 273
PBX SETTINGS ..................................................................................... 274
General Settings ............................................................................................................................... 274
Voice Prompt Customization ............................................................................................................ 276
Record New Custom Prompt .................................................................................................... 276
Download All Custom Prompt ................................................................................................... 277
PBX Settings/Jitter Buffer ................................................................................................................. 278
PBX Settings/RTP Settings .............................................................................................................. 278
PBX Settings/Payload ...................................................................................................................... 279
IAX SETTINGS ....................................................................................... 280
IAX Settings/General ........................................................................................................................ 280
IAX Settings/Registration .................................................................................................................. 280
IAX Settings/Security ........................................................................................................................ 281
SIP SETTINGS ....................................................................................... 282
SIP Settings/General ........................................................................................................................ 282
SIP Settings/MISC ............................................................................................................................ 282
SIP Settings/Session Timer .............................................................................................................. 283
SIP Settings/TCP and TLS ............................................................................................................... 284
SIP Settings/NAT .............................................................................................................................. 285
SIP Settings/TOS.............................................................................................................................. 285
SIP Settings/SIP Trunk Prompt Tone................................................................................................ 287
Transparent Call-Info header ............................................................................................................ 288
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INTERFACE SETTINGS ......................................................................... 289
CTI SERVER .......................................................................................... 292
ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) ............ 293
CRM INTEGRATION .............................................................................. 294
SugarCRM ........................................................................................................................................ 294
Salesforce CRM ............................................................................................................................... 295
PMS INTEGRATION............................................................................... 298
HMobile PMS Connector .................................................................................................................. 298
Mitel PMS ......................................................................................................................................... 299
Configuration .................................................................................................................................... 300
Basic Settings ........................................................................................................................... 300
PMS Features ................................................................................................................................... 300
Room Status ............................................................................................................................. 300
Wake Up Service ...................................................................................................................... 302
Mini Bar ..................................................................................................................................... 303
WAKEUP SERVICE ............................................................................... 306
WakeUp Service using Admin Login ................................................................................................ 306
WakeUp Service from User Portal .................................................................................................... 307
WakeUp Service using Feature Code .............................................................................................. 307
ANNOUNCEMENTS CENTER ............................................................... 308
Announcements Center Settings ...................................................................................................... 309
Group Settings .................................................................................................................................. 309
STATUS AND REPORTING ................................................................... 312
PBX Status ....................................................................................................................................... 312
Trunks ....................................................................................................................................... 313
Extensions................................................................................................................................. 314
Interfaces Status ....................................................................................................................... 315
System Status .................................................................................................................................. 316
General ..................................................................................................................................... 316
Network ..................................................................................................................................... 316
Storage Usage .......................................................................................................................... 317
Resource Usage ....................................................................................................................... 318
System Events .................................................................................................................................. 319
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Alert Events List ........................................................................................................................ 319
Alert Log .................................................................................................................................... 321
Alert Contact ............................................................................................................................. 322
CDR .................................................................................................................................................. 322
CDR Improvement .................................................................................................................... 327
Downloaded CDR File .............................................................................................................. 328
Statistics .................................................................................................................................... 329
Recording Files ......................................................................................................................... 330
API Configuration ...................................................................................................................... 331
USER PORTAL ...................................................................................... 332
Basic Information .............................................................................................................................. 334
Personal Data ................................................................................................................................... 334
Value-added Features ...................................................................................................................... 334
MAINTENANCE ..................................................................................... 335
User Management ............................................................................................................................ 335
User Information ....................................................................................................................... 335
Custom Privilege ....................................................................................................................... 336
Concurrent Multi-User Login ..................................................................................................... 338
Change Password ..................................................................................................................... 339
Change binding Email ............................................................................................................... 340
Login Settings ........................................................................................................................... 340
Operation Log ........................................................................................................................... 341
Upgrading ......................................................................................................................................... 343
Upgrading Via Network ............................................................................................................. 343
Upgrading Via Local Upload ..................................................................................................... 345
No Local Firmware Servers ...................................................................................................... 346
Backup .............................................................................................................................................. 347
Backup/Restore ........................................................................................................................ 347
Data Sync.................................................................................................................................. 349
Restore Configuration from Backup File ................................................................................... 351
System Cleanup/Reset ..................................................................................................................... 352
Reset and Reboot ..................................................................................................................... 352
Cleaner ...................................................................................................................................... 352
USB/SD Card Files Cleanup ..................................................................................................... 354
Syslog ............................................................................................................................................... 355
Network Troubleshooting .................................................................................................................. 356
Ethernet Capture ....................................................................................................................... 356
IP Ping ....................................................................................................................................... 357
Traceroute ................................................................................................................................. 358
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Signaling Troubleshooting ................................................................................................................ 358
Analog Record Trace ................................................................................................................ 358
Service Check .................................................................................................................................. 360
Network Status ................................................................................................................................. 360
EXPERIENCING THE UCM6200 SERIES IP PBX ................................. 362
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Table of Tables
Table 1: Technical Specifications ........................................................................................................................ 28
Table 2: UCM6200 Equipment Packaging .......................................................................................................... 31
Table 3: LCD Menu Options ................................................................................................................................ 36
Table 4: UCM6202/UCM6204 LED Indicators .................................................................................................... 37
Table 5: UCM6208 LED Indicators ...................................................................................................................... 37
Table 6: UCM6200 Network SettingsBasic Settings........................................................................................ 43
Table 7: UCM6200 Network Settings802.1X ................................................................................................... 51
Table 8: UCM6200 Network SettingsStatic Routes ......................................................................................... 51
Table 9: UCM6200 Network SettingsPort Forwarding ..................................................................................... 53
Table 10: UCM6200 System SettingsNetwork SettingsOpen VPN ............................................................. 56
Table 11: UCM6200 FirewallStatic DefenseCurrent Service ....................................................................... 59
Table 12: Typical Firewall Settings ...................................................................................................................... 60
Table 13: Firewall Rule Settings .......................................................................................................................... 61
Table 14: UCM6200 Firewall Dynamic Defense ................................................................................................. 62
Table 15: Fail2Ban Settings ................................................................................................................................ 64
Table 16: HTTP Server Settings .......................................................................................................................... 74
Table 17: Time Auto Updating ............................................................................................................................. 75
Table 18: Create New Office Time ...................................................................................................................... 77
Table 19: Create New Holiday............................................................................................................................. 79
Table 20: Email Settings ...................................................................................................................................... 80
Table 21: Email Log ............................................................................................................................................. 83
Table 22: Auto Provision Settings ....................................................................................................................... 92
Table 23: Global Policy ParametersLocalization ............................................................................................. 97
Table 24: Global Policy ParametersPhone Settings ........................................................................................ 97
Table 25: Global Policy ParametersContact List ............................................................................................. 98
Table 26: Global Policy ParametersMaintenance.......................................................................................... 100
Table 27: Global Policy ParametersNetwork Settings ................................................................................... 102
Table 28: Global Policy ParametersCustomization ....................................................................................... 102
Table 29: Create New Template ........................................................................................................................ 104
Table 30: Create New Model Template ............................................................................................................. 105
Table 31: SIP Extension Configuration ParametersBasic Settings ............................................................... 122
Table 32: SIP Extension Configuration ParametersMedia ............................................................................ 123
Table 33: SIP Extension Configuration ParametersFeatures ........................................................................ 124
Table 34: SIP Extension Configuration ParametersSpecific Time ................................................................. 129
Table 35: IAX Extension Configuration ParametersBasic Settings ............................................................... 129
Table 36: IAX Extension Configuration ParametersMedia ............................................................................ 130
Table 37: IAX Extension Configuration ParametersFeatures ........................................................................ 131
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Table 38: IAX Extension Configuration ParametersSpecific Time................................................................. 133
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Table 39: FXS Extension Configuration ParametersBasic Settings .............................................................. 133
Table 40: FXS Extension Configuration ParametersMedia ........................................................................... 134
Table 41: FXS Extension Configuration ParametersFeatures ....................................................................... 135
Table 42: FXS Extension Configuration ParametersSpecific Time ............................................................... 138
Table 43: Batch Add SIP Extension Parameters ............................................................................................... 138
Table 44: Batch Add IAX Extension Parameters ............................................................................................... 141
Table 45: SIP extensions Imported File Example ............................................................................................. 146
Table 46: IAX extensions Imported File Example ............................................................................................. 148
Table 47: FXS Extensions Imported File Example ............................................................................................ 150
Table 48: Analog Trunk Configuration Parameters ........................................................................................... 158
Table 49: PSTN Detection for Analog Trunk ..................................................................................................... 163
Table 50: Create New SIP Trunk ....................................................................................................................... 164
Table 51: SIP Register Trunk Configuration Parameters .................................................................................. 165
Table 52: SIP Peer Trunk Configuration Parameters ........................................................................................ 169
Table 53: Create New IAX Trunk ....................................................................................................................... 171
Table 54: IAX Register Trunk Configuration Parameters .................................................................................. 171
Table 55: IAX Peer Trunk Configuration Parameters ........................................................................................ 173
Table 56: SLA Station Configuration Parameters ............................................................................................. 176
Table 57: Outbound Route Configuration Parameters ...................................................................................... 179
Table 58: Outbound Routes/PIN Group ............................................................................................................ 183
Table 59: Inbound Rule Configuration Parameters ........................................................................................... 187
Table 60: Conference Room Configuration Parameters ................................................................................... 194
Table 61: Conference Settings .......................................................................................................................... 196
Table 62: Conference Caller IVR Menu ............................................................................................................ 198
Table 63: Conference Schedule Parameters .................................................................................................... 200
Table 64: IVR Configuration Parameters .......................................................................................................... 205
Table 65: Voicemail Settings ............................................................................................................................. 214
Table 66: Voicemail IVR Menu .......................................................................................................................... 215
Table 67: Voicemail Email Settings ................................................................................................................... 217
Table 68: Voicemail Group Settings .................................................................................................................. 218
Table 69: Ring Group Parameters .................................................................................................................... 219
Table 70: Paging/Intercom Group Configuration Parameters ........................................................................... 225
Table 71: Call Queue Configuration Parameters .............................................................................................. 226
Table 72: Static Agent Limitation ....................................................................................................................... 229
Table 73: Call Center Parameters ..................................................................................................................... 231
Table 74: Switchboard Parameters ................................................................................................................... 233
Table 75: FAX/T.38 Settings .............................................................................................................................. 241
Table 76: SIP Presence Status ......................................................................................................................... 249
Table 77: Follow Me Settings ............................................................................................................................ 251
Table 78: Follow Me Options ............................................................................................................................. 251
Table 79: DISA Settings .................................................................................................................................... 254
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Table 80: Callback Configuration Parameters................................................................................................... 255
Table 81: Event List Settings ............................................................................................................................. 256
Table 82: UCM6200 Feature Codes ................................................................................................................. 266
Table 83: Internal Options/General ................................................................................................................... 274
Table 84: Internal Options/Jitter Buffer .............................................................................................................. 278
Table 85: Internal Options/RTP Settings ........................................................................................................... 278
Table 86: Internal Options/Payload ................................................................................................................... 279
Table 87: IAX Settings/General ......................................................................................................................... 280
Table 88: IAX Settings/Registration .................................................................................................................. 280
Table 89: IAX Settings/Static Defense .............................................................................................................. 281
Table 90: SIP Settings/General ......................................................................................................................... 282
Table 91: SIP Settings/Misc .............................................................................................................................. 282
Table 92: SIP Settings/Session Timer ............................................................................................................... 283
Table 93: SIP Settings/TCP and TLS ................................................................................................................ 284
Table 94: SIP Settings/NAT ............................................................................................................................... 285
Table 95: SIP Settings/ToS................................................................................................................................ 285
Table 96: PBX Interface Settings ...................................................................................................................... 289
Table 97: SugarCRM Settings ........................................................................................................................... 294
Table 98: Salesforce Settings............................................................................................................................ 296
Table 99: PMS Supported Features .................................................................................................................. 299
Table 100: PMS Basic Settings ......................................................................................................................... 300
Table 101: PMS Wake up Service ..................................................................................................................... 302
Table 102: Create New Mini Bar ....................................................................................................................... 303
Table 103: Create New Maid ............................................................................................................................. 304
Table 104: Wakeup Service .............................................................................................................................. 307
Table 105: Announcements Center Settings ..................................................................................................... 309
Table 106: Group Settings................................................................................................................................. 309
Table 107: Trunk Status .................................................................................................................................... 313
Table 108: Extension Status.............................................................................................................................. 314
Table 109: Interface Status Indicators ............................................................................................................... 315
Table 110: System StatusGeneral ................................................................................................................. 316
Table 111: System StatusNetwork ................................................................................................................. 317
Table 112: CDR Filter Criteria ........................................................................................................................... 323
Table 113: CDR Statistics Filter Criteria ............................................................................................................ 330
Table 114: API Configuration Files .................................................................................................................... 331
Table 115: User ManagementCreate New User ............................................................................................ 336
Table 116: Change Binding Email option .......................................................................................................... 340
Table 117: Operation Log Column Header ........................................................................................................ 342
Table 118: Network Upgrade Configuration ...................................................................................................... 344
Table 119: Data Sync Configuration .................................................................................................................. 350
Table 120: Cleaner Configuration ..................................................................................................................... 353
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Table 121: USB/SD Card Files Cleanup ........................................................................................................... 355
Table 122: Ethernet Capture ............................................................................................................................. 357
Table of Figures
Figure 1: UCM6202 Front View........................................................................................................................... 31
Figure 2: UCM6202 Back View ........................................................................................................................... 32
Figure 3: UCM6204 Front View........................................................................................................................... 32
Figure 4: UCM6204 Back View ........................................................................................................................... 33
Figure 5: UCM6208 Front View........................................................................................................................... 34
Figure 6: UCM6208 Back View ........................................................................................................................... 34
Figure 7: UCM6202 Web GUI Login Page .......................................................................................................... 38
Figure 8: Default Random Password .................................................................................................................. 39
Figure 9: UCM6200 Setup Wizard ...................................................................................................................... 40
Figure 10: UCM6200 Web GUI Language .......................................................................................................... 41
Figure 11: UCM6202 Network Interface Method: Route ..................................................................................... 47
Figure 12: UCM6202 Network Interface Method: Switch .................................................................................... 47
Figure 13: UCM6202 Network Interface Method: Dual ....................................................................................... 48
Figure 14: DHCP Client List ................................................................................................................................ 48
Figure 15: Add MAC Address Bind ..................................................................................................................... 49
Figure 16: Batch Add MAC Address Bind ........................................................................................................... 49
Figure 17: UCM6200 Using 802.1X as Client ..................................................................................................... 50
Figure 18: UCM6200 Using 802.1X EAP-MD5 ................................................................................................... 50
Figure 19: UCM6204 Static Route Sample ......................................................................................................... 52
Figure 20: UCM6204 Static Route Configuration................................................................................................ 53
Figure 21: Create New Port Forwarding ............................................................................................................. 54
Figure 22: UCM6200 Port Forwarding Configuration ......................................................................................... 55
Figure 23: GXP2160 Web Access using UCM6202 port forwarding .................................................................. 55
Figure 24: Open VPN feature on the UCM6200 ................................................................................................. 56
Figure 25: Register Domain Name on noip.com ................................................................................................. 57
Figure 26: UCM6200 DDNS Setting ................................................................................................................... 58
Figure 27: Using Domain Name to Connect to UCM6200 .................................................................................. 58
Figure 28: Create New Firewall Rule .................................................................................................................. 61
Figure 29: Configure Dynamic Defense .............................................................................................................. 63
Figure 30: Fail2ban Settings ............................................................................................................................... 64
Figure 31: SSH Access ....................................................................................................................................... 65
Figure 32: LDAP Server Configurations .............................................................................................................. 67
Figure 33: Default LDAP Phonebook DN ............................................................................................................ 67
Figure 34: Default LDAP Phonebook Attributes .................................................................................................. 68
Figure 35: LDAP ServerLDAP Phonebook ...................................................................................................... 68
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Figure 36: Add LDAP Phonebook ....................................................................................................................... 69
Figure 37: Edit LDAP Phonebook ....................................................................................................................... 69
Figure 38: Import Phonebook.............................................................................................................................. 69
Figure 39: Phonebook CSV File Format ............................................................................................................. 70
Figure 40: LDAP Phonebook After Import ........................................................................................................... 70
Figure 41: Export Selected LDAP Phonebook .................................................................................................... 71
Figure 42: LDAP Client Configurations ............................................................................................................... 72
Figure 43: GXP2170 LDAP Phonebook Configuration ....................................................................................... 73
Figure 44: Set Time Manually ............................................................................................................................. 76
Figure 45: Create New Office Time ..................................................................................................................... 77
Figure 46: SettingsTime SettingsOffice Time ............................................................................................... 78
Figure 47: Create New Holiday ........................................................................................................................... 78
Figure 48: SettingsTime SettingsHoliday ..................................................................................................... 79
Figure 49: UCM6200 Email Settings ................................................................................................................... 81
Figure 50: Email Templates ................................................................................................................................. 82
Figure 51: Conference Schedule Template ......................................................................................................... 82
Figure 52: Email Send log ................................................................................................................................... 83
Figure 53: Email Logs ......................................................................................................................................... 84
Figure 54: SettingsRecordings Storage .......................................................................................................... 85
Figure 55: Recordings Storage Prompt Information ........................................................................................... 85
Figure 56: Recording Storage Category ............................................................................................................. 86
Figure 57: Google Service SettingsOAuth2.0 Authentication .......................................................................... 87
Figure 58: Google ServiceNew Project ........................................................................................................... 87
Figure 59: Google ServiceCreate New Credential .......................................................................................... 88
Figure 60: Google ServiceOAuth2.0 Login ...................................................................................................... 88
Figure 61: Zero Config Configuration Architecture for End Point Device ........................................................... 90
Figure 62: UCM6200 Zero Config ....................................................................................................................... 91
Figure 63: Auto Provision Settings ...................................................................................................................... 92
Figure 64: Auto Discover ..................................................................................................................................... 94
Figure 65: Discovered Devices ........................................................................................................................... 94
Figure 66: Device list - CSV file sample .............................................................................................................. 95
Figure 67: Managing Discovered Devices .......................................................................................................... 95
Figure 68: Global Policy Categories ................................................................................................................... 96
Figure 69: Edit Global Template ........................................................................................................................ 104
Figure 70: Edit Model Template ........................................................................................................................ 107
Figure 71: Template Management .................................................................................................................... 108
Figure 72: Upload Model Template Manually .................................................................................................... 108
Figure 73: Create New Device .......................................................................................................................... 109
Figure 74: Manage Devices .............................................................................................................................. 110
Figure 75: Edit Device ....................................................................................................................................... 110
Figure 76: Edit Customize Device Settings ....................................................................................................... 112
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Figure 77: Add P Value in Customize Device Settings ..................................................................................... 113
Figure 78: Modify Selected Devices - Same Model .......................................................................................... 114
Figure 79: Modify Selected Devices - Different Models .................................................................................... 115
Figure 80: Device List in Zero Config ................................................................................................................ 116
Figure 81: Zero Config Sample - Global Policy................................................................................................. 117
Figure 82: Zero Config Sample - Device Preview 1 .......................................................................................... 118
Figure 83: Zero Config Sample - Device Preview 2 .......................................................................................... 119
Figure 84: Zero Config Sample - Device Preview 3 .......................................................................................... 120
Figure 85: Create New Device .......................................................................................................................... 121
Figure 86: Manage Extensions ......................................................................................................................... 143
Figure 87: Export Extensions ............................................................................................................................ 144
Figure 88: Import Extensions ............................................................................................................................ 145
Figure 89: Import File ........................................................................................................................................ 145
Figure 90: Import Error ...................................................................................................................................... 151
Figure 91: E-mail Notification - Prompt Information .......................................................................................... 152
Figure 92: Account Registration Information and QR Code .............................................................................. 153
Figure 93: LDAP Client Information and QR Code ........................................................................................... 153
Figure 94: Multiple Registrations per Extension ............................................................................................... 154
Figure 95: Extension - Concurrent Registration ................................................................................................ 154
Figure 96: SMS Message Support .................................................................................................................... 155
Figure 97: Edit Extension Group ....................................................................................................................... 156
Figure 98: Select Extension Group in Outbound Route .................................................................................... 157
Figure 99: UCM6200 FXO Tone Settings ......................................................................................................... 161
Figure 100: UCM6200 PSTN Detection ............................................................................................................ 161
Figure 101: UCM6200 PSTN Detection: Auto Detect ....................................................................................... 162
Figure 102: UCM6200 PSTN Detection: Semi-Auto Detect ............................................................................. 162
Figure 103: DOD extension selection ............................................................................................................... 174
Figure 104: Edit DOD ........................................................................................................................................ 175
Figure 105: SLA Station .................................................................................................................................... 176
Figure 106: Enable SLA Mode for Analog Trunk ............................................................................................... 177
Figure 107: Analog Trunk with SLA Mode Enabled .......................................................................................... 177
Figure 108: SLA Example - SLA Station ........................................................................................................... 178
Figure 109: SLA Example - MPK Configuration ................................................................................................ 178
Figure 110: Country Codes ............................................................................................................................... 182
Figure 111: Create New PIN Group .................................................................................................................. 183
Figure 112: PIN Members ................................................................................................................................. 184
Figure 113: Outbound PIN ................................................................................................................................ 184
Figure 114: CDR Record ................................................................................................................................... 185
Figure 115: Importing PIN Groups from CSV files ............................................................................................ 185
Figure 116: Incorrect CSV File .......................................................................................................................... 186
Figure 117: CSV File Format ............................................................................................................................. 186
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Figure 118: CSV File Successful Upload .......................................................................................................... 187
Figure 119: Inbound Route feature: Prepend .................................................................................................... 190
Figure 120: Inbound Route - Multiple Mode...................................................................................................... 190
Figure 121: Inbound Route - Multiple Mode Feature Codes............................................................................. 191
Figure 122: Blacklist Configuration Parameters................................................................................................ 192
Figure 123: Blacklist csv File............................................................................................................................. 193
Figure 124: Conference .................................................................................................................................... 196
Figure 125: Conference Invitation from Web GUI ............................................................................................. 197
Figure 126: Conference Recording ................................................................................................................... 199
Figure 127: Conference Schedule .................................................................................................................... 203
Figure 128: Create New IVR ............................................................................................................................. 204
Figure 129: Key Pressing Events ...................................................................................................................... 207
Figure 130: Black/White List ............................................................................................................................. 208
Figure 131: Click on Prompt to Create IVR Prompt .......................................................................................... 209
Figure 132: Language Settings for Voice Prompt ............................................................................................. 210
Figure 133: Voice Prompt Package List ............................................................................................................ 211
Figure 134: New Voice Prompt Language Added ............................................................................................. 211
Figure 135: Upload Single Voice Prompt for Entire Language Pack ................................................................ 212
Figure 136: Voicemail Settings.......................................................................................................................... 213
Figure 137: Voicemail Count ............................................................................................................................. 216
Figure 138: Voicemail Email Settings ............................................................................................................... 217
Figure 139: Voicemail Group............................................................................................................................. 218
Figure 140: Ring Group ..................................................................................................................................... 219
Figure 141: Ring Group Configuration .............................................................................................................. 221
Figure 142: Sync LDAP Server option .............................................................................................................. 222
Figure 143: Manually Sync LDAP Server ......................................................................................................... 223
Figure 144: Ring Group Remote Extension ...................................................................................................... 223
Figure 145: Paging/Intercom Group .................................................................................................................. 224
Figure 146: Page/Intercom Group Settings ...................................................................................................... 225
Figure 147: Call Queue ..................................................................................................................................... 226
Figure 148: Static Agents limitation ................................................................................................................... 229
Figure 149: Agent Login Settings ...................................................................................................................... 230
Figure 150: Call Queue Statistics ..................................................................................................................... 232
Figure 151: Call Queue Switchboard ................................................................................................................ 233
Figure 152: Edit Pickup Group .......................................................................................................................... 235
Figure 153: Edit Pickup Feature Code .............................................................................................................. 236
Figure 154: Music On Hold Default Class ......................................................................................................... 237
Figure 155: Play Custom Prompt ...................................................................................................................... 238
Figure 156: Information Prompt ........................................................................................................................ 238
Figure 157: Record Custom Prompt ................................................................................................................. 239
Figure 158: Fax Settings ................................................................................................................................... 240
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Figure 159: Configure Analog Trunk without Fax Detection ............................................................................. 242
Figure 160: Configure Extension for Fax Machine: FXS Extension ................................................................. 243
Figure 161: Configure Extension for Fax Machine: Analog Settings ................................................................ 243
Figure 162: Configure Inbound Rule for Fax ..................................................................................................... 244
Figure 163: Create Fax Extension .................................................................................................................... 244
Figure 164: Inbound Route to Fax Extension ................................................................................................... 245
Figure 165: List of Fax Files .............................................................................................................................. 245
Figure 166: Fax Sending in Web GUI ............................................................................................................... 246
Figure 167: Fax Send Progress ........................................................................................................................ 246
Figure 168: SIP Presence Configuration .......................................................................................................... 248
Figure 169: SIP Presence Feature Code .......................................................................................................... 249
Figure 170: Edit Follow Me ............................................................................................................................... 250
Figure 171: Speed Dial Destinations ................................................................................................................ 252
Figure 172: Create New DISA........................................................................................................................... 253
Figure 173: Create New Event List ................................................................................................................... 257
Figure 174: Create Dial by Name Group .......................................................................................................... 259
Figure 175: Configure Extension First Name and Last Name .......................................................................... 260
Figure 176: Dial By Name Group In IVR Key Pressing Events ........................................................................ 261
Figure 177: Dial By Name Group In Inbound Rule ........................................................................................... 261
Figure 178: StatusPBX StatusActive Calls - Ringing ................................................................................. 263
Figure 179: StatusPBX StatusActive Calls – Call Established .................................................................. 263
Figure 180: Configure to Monitor an Active Call ............................................................................................... 264
Figure 181: Enable/Disable Feature codes ....................................................................................................... 271
Figure 182: Download Recording File from CDR Page .................................................................................... 272
Figure 183: Download Recording File from Recording Files Page ................................................................... 272
Figure 184: Record New Custom Prompt ......................................................................................................... 276
Figure 185: Upload Custom Prompt ................................................................................................................. 277
Figure 186: Download All Custom Prompt ........................................................................................................ 277
Figure 187: SIP Trunk Prompt Tone .................................................................................................................. 287
Figure 188: Transparent Call-Info ..................................................................................................................... 288
Figure 189: FXS Ports Signaling Preference .................................................................................................... 289
Figure 190: FXO Ports ACIM Settings .............................................................................................................. 289
Figure 191: DAHDI Settings .............................................................................................................................. 291
Figure 192: CTI Server Listening port ............................................................................................................... 292
Figure 193: SugarCRM Basic Settings ............................................................................................................. 294
Figure 194: CRM User Settings ........................................................................................................................ 295
Figure 195: Salesforce Basic Settings .............................................................................................................. 296
Figure 196: Salesforce User Settings ............................................................................................................... 297
Figure 197: UCM & PMS interaction ................................................................................................................. 298
Figure 198: UCM & PMS interaction ................................................................................................................. 299
Figure 199: Create New Room ......................................................................................................................... 301
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Figure 200: Room Status .................................................................................................................................. 301
Figure 201: Add batch rooms ............................................................................................................................ 302
Figure 202: Create New Wake Up Service ....................................................................................................... 302
Figure 203: Wakeup Call executed ................................................................................................................... 303
Figure 204: Create New Mini Bar ...................................................................................................................... 303
Figure 205: Create New Maid ........................................................................................................................... 304
Figure 206: Create New Consumer Goods....................................................................................................... 304
Figure 207: Mini Bar .......................................................................................................................................... 305
Figure 208: Create New Wakeup Service ......................................................................................................... 306
Figure 209: Announcements Center ................................................................................................................. 308
Figure 210: Announcements Center Group Configuration ................................................................................ 310
Figure 211: Announcements Center Code Configuration ................................................................................. 310
Figure 212: Announcements Center Example .................................................................................................. 311
Figure 213: StatusPBX Status ....................................................................................................................... 312
Figure 214: Trunk Status ................................................................................................................................... 313
Figure 215: Extension Status ............................................................................................................................ 314
Figure 216: UCM6204 Interfaces Status ........................................................................................................... 315
Figure 217: System StatusStorage Usage .................................................................................................... 318
Figure 218: System StatusResource Usage ................................................................................................. 318
Figure 219: System EventsAlert Events Lists: Disk Usage ........................................................................... 319
Figure 220: System EventsAlert Events Lists: External Disk Usage............................................................. 320
Figure 221: System EventsAlert Events Lists: Memory Usage ..................................................................... 320
Figure 222: System EventsAlert Events Lists: System Crash....................................................................... 321
Figure 223: System EventsAlert Log ............................................................................................................. 321
Figure 224: Filter for Alert Log .......................................................................................................................... 322
Figure 225: CDR Filter ...................................................................................................................................... 323
Figure 226: Call Report ..................................................................................................................................... 325
Figure 227: Call Report Entry with Audio Recording File .................................................................................. 326
Figure 228: Automatic Download Settings ........................................................................................................ 327
Figure 229: CDR Report ................................................................................................................................... 327
Figure 230: Detailed CDR Information .............................................................................................................. 327
Figure 231: Downloaded CDR File Sample ...................................................................................................... 328
Figure 232: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ........................................ 328
Figure 233: Downloaded CDR File Sample - Source Channel and Dest Channel 2 ........................................ 328
Figure 234: CDR Statistics ................................................................................................................................ 329
Figure 235: CDRRecording Files................................................................................................................... 330
Figure 236: Edit User Information by Super Admin .......................................................................................... 332
Figure 237: User Portal Login ........................................................................................................................... 333
Figure 238: User Portal Layout ......................................................................................................................... 333
Figure 239: User Management Page Display ................................................................................................... 335
Figure 240: Create New User ........................................................................................................................... 335
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Figure 241: User Management – New Users.................................................................................................... 336
Figure 242: General User.................................................................................................................................. 337
Figure 243: Create New Custom Privilege ........................................................................................................ 338
Figure 244: Multiple User Operation Error Prompt ........................................................................................... 339
Figure 245 : Change Password ........................................................................................................................ 339
Figure 246: Change Binding Email ................................................................................................................... 340
Figure 247: Login Timeout Settings .................................................................................................................. 341
Figure 248: Operation Logs .............................................................................................................................. 342
Figure 249: Operation Logs Filter ..................................................................................................................... 343
Figure 250: Network Upgrade ........................................................................................................................... 344
Figure 251: Local Upgrade................................................................................................................................ 345
Figure 252: Upgrading Firmware Files .............................................................................................................. 345
Figure 253: Reboot UCM6200 .......................................................................................................................... 346
Figure 254: Create New Backup ....................................................................................................................... 347
Figure 255: Restore Warning ............................................................................................................................ 348
Figure 256: Backup / Restore ........................................................................................................................... 348
Figure 257: Local Backup ................................................................................................................................. 349
Figure 258: Data Sync ...................................................................................................................................... 350
Figure 259: Restore UCM6200 from Backup File ............................................................................................. 351
Figure 260: Reset and Reboot .......................................................................................................................... 352
Figure 261: Cleaner .......................................................................................................................................... 353
Figure 262: USB/SD Card Files Cleanup.......................................................................................................... 355
Figure 263: Ethernet Capture............................................................................................................................ 356
Figure 264: Ping ................................................................................................................................................ 357
Figure 265: Traceroute ...................................................................................................................................... 358
Figure 266: Troubleshooting Analog Trunks ..................................................................................................... 359
Figure 267: A Key Dial-up FXO ......................................................................................................................... 360
Figure 268: Service Check ................................................................................................................................ 360
Figure 269: Network Status............................................................................................................................... 361
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DOCUMENT PURPOSE
This document describes UCM6200 series specifications, features and will help you to configure your system
via Web GUI menu to fully manipulate the supported features. The intended audiences of this document are
device
administrators.
To
learn
more
about
UCM6200
series
features,
please
http://www.grandstream.com/support to download available how-to guides.
This guide covers following main topics:
•
Product overview
•
Busy camp-on
•
Installation
•
Presence
•
Getting started
•
Follow me
•
System settings
•
Speed Dial
•
Provisioning
•
DISA
•
Extensions
•
Callback
•
Extension groups
•
BLF and event list
•
Analog Trunks
•
Dial by name
•
VoIP Trunks
•
Active calls and monitor
•
SLA station
•
Call features
•
Call routes
•
Call recording
•
Conference room.
•
CTI Server
•
Conference schedule
•
Asterisk manager interface (AMI)
•
IVR
•
CRM integration
•
Language settings for voice prompt
•
PMS integration
•
Voicemail
•
Wakeup service
•
Ring group
•
Announcements center
•
Paging and intercom group
•
Status and reporting
•
Call queue
•
CDR (Call Details Record)
•
Pickup groups
•
User Portal
•
PIN Groups
•
Upgrading and maintenance
•
Music on hold
•
Backup/restore
•
Fax Server
•
Troubleshooting
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visit
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 1.0.15.14
•
No major changes.
Firmware Version 1.0.15.13
•
Added support to announce name in Dial By Name feature. [DIAL BY NAME]
•
Increased maximum number of call queue static agents. [Static Agents limitation]
•
Added Queue Log Cleaner. [Queue Log Cleaner]
•
Added operation logs details and remarks. [Operation Log]
•
Added extension level voicemail-to-Email setting. [Send Voicemail to Email] [Keep Voicemail after
Emailing]
•
Added support for reset certificate. [Reset Certificates]
•
Added support for GXP21xx color phone queue login/logout softkey. [Enable Agent Login]
•
Added support to add comments to inbound/outbound route patterns. [Pattern] [Pattern]
•
Added PPI mode option under SIP trunk advanced settings [PPI Mode]
•
Added ability to import PIN groups from CSV files. [Importing PIN Groups from CSV files]
•
Added support for wakeup groups. [WAKEUP SERVICE]
•
Added support for SYN Flood defense. [SYN-Flood Defense Enable]
•
Added Queue Log option to Backup/Restore page. [Backup/Restore]
•
Modified Switchboard UI to offer easier access to call Options. [Switchboard]
•
Further optimized Call Queue Statistics page to provide a more user-friendly experience and improve
performance. [Queue Statistics]
•
Added image uploading support to Email templates [Email Templates]
•
Added IPv6 gateway support. [Static Routes]
•
Improved Voice Message Responses for failed SIP trunk calls. [SIP Trunk Prompt Tone]
•
Added Fail2Ban to support TCP/TLS beside the already-supported UDP. [Fail2ban]
•
Restored ability to sort the ringing order of Follow Me numbers. [FOLLOW ME]
•
Added option to restrict users from changing their SIP credentials through user portal [Consumer]
•
Added option to enable or disable fax-to-mail feature on extension level. [Fax To Email]
•
Added IP address whitelist to web GUI configuration. [Enable IP whitelist]
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Firmware Version 1.0.14.23
•
Restored ability to view voicemail count in the Extension/Trunk overview. [Extension Voicemail Count]
•
Restored the ability to set custom numbers for call forwarding settings. [Call Forward Unconditional]
[Call Forward No Answer] [Call Forward Busy]
•
Restored previous format for entering multiple dial plans (one pattern per line) for inbound/outbound rules.
[Pattern] [Pattern]
•
Restored Zero Config’s sorting by column and introduced a search bar. [Managing discovered devices]
Firmware Version 1.0.14.21
•
A new Web GUI consistent operating style, can display/update the system's status in real time.
•
Added support for SIP Presence. [PRESENCE]
•
Added support for Call Center feature/Virtual Call Queue. [Call Center Settings and enhancements]
•
Added support for Call Queue position announcement. [Call Center Settings and enhancements]
•
Added support for Call Queue Statistics. [Queue Statistics]
•
Added support for Call Queue Auto-Fill. [Queue Auto fill enhancement]
•
Added switchboard for call queue monitoring. [Switchboard]
•
Added ability to restore blind transfer call to transferrer. [Allow callback when blind transfer fails]
•
Added support for external disk cleaner. [System Cleanup/Reset]
•
Added option to enable DOD when call is being diverted/forwarded. [Use callee DOD on FWD or Ring
Simultaneously]
•
Change follow me settings to extension level settings. [FOLLOW ME]
•
Added support for call forward whitelist. [FWD Whitelist]
•
Added Fail2Ban defense from web login attack. [Login Attack Defense]
•
Added limitation for maximum number of call queue static agents. [Static Agents limitation]
•
Added support for wakeup service module in Custom privilege. [Custom Privilege]
•
Added IPv6 support for T.38.
•
Added DAHDI settings. [DAHDI Settings]
•
Added ability to pass through SIP Call-Info header to support GXP phone JPEG_Over_HTTP with
encryption and authentication to open door for GDS3710. [Transparent Call-Info header]
Firmware Version 1.0.13.14
•
Added extension whitelist/blacklist for IVR dialing. [IVR]
•
Added ability to include DOD in PPI Header for SIP trunk. [PPI Mode]
•
Added advanced IPv6 support including IPv6-to-IPv4 SIP calls, IPv6 router, IPv6 iptables/Static defense.
•
Added ability to customize PAI Header. [PAI Header]
•
Added blacklist for outbound calls. [Outbound Blacklist]
•
Added support to upload/download MOH package from Web GUI. [MUSIC ON HOLD]
•
Added support to download custom prompts from Web GUI. [Download All Custom Prompt]
•
Added option to configure prompt timeout in Dial By Name. [DIAL BY NAME]
•
Added description field in ZeroConfig settings to configure Softkey/Line/MPK for GXP series phones.
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[PROVISIONING]
•
Improved seamless transfer privilege control. [Seamless transfer privilege control]
•
Added RTP Keep-alive support. [RTP Keep-alive]
•
Added Email Send Log. [Email Send Log]
•
Added support for Mitel simulation/protocol interfaces for PMS module. [PMS]
•
Added support for up to 10 failover trunks. [Use Failover Trunk]
Firmware Version 1.0.12.19
•
Added support for binding a mobile phone number to extension. [Mobile Phone Number]
•
Added support OPUS codec.
•
Added support call-barging privilege settings based on extensions. [Monitor privilege control]
•
Added support for Seamless Transfer. [Seamless Transfer]
•
Added support for Custom Call-Info for Auto Answer. [Custom Call-info for Auto Answer]
•
Added support for DND Whitelist. [Do Not Disturb]
•
Add the Field Description on Softkey, Line keys and MPK from Zero Config.
•
Added support to select interval for numbers on Batch add extension. [Extension Interval]
•
Added support for Batch Add CallerID Number. [CallerID Number]
•
Added support for Search Extensions Using CallerID Name.
•
Added support to Enable/Disable Inbound and Outbound Route. [Disable This Route/Disable This Route]
•
Added support for Outbound Route Time Condition. [Time Condition]
•
Added support for IPv6. [IPv6 Address]
•
Added Support for MTU configurable. [MTU]
•
Added support of CRM. [CRM]
•
Added support for Custom Privilege in User Management. [Custom Privilege]
•
Added Hotline support for FXS Extension. [Hotline]
•
Added support for Separate Wakeup Service. [WAKEUP SERVICE]
•
Added ability to provision phones from different network subnets using zero config. [Subnet Whitelist]
•
One-key-dial is replaced by Speed Dial to support more than one digit. [SPEED DIAL]
•
Added Append extension number in the end of DOD. [Direct Outward Dialing (DOD)]
•
Support Japan CID NTT Detect.
•
Added support for Ethernet Capture Auto Sync to SFTP Server. [Enable SFTP Data Sync]
•
Added support for Ethernet Capture saved to External Storage Device. [Storage to External Device]
•
Added support for Disable Extension Range on the Setup Wizard. [Setup Wizard]
•
Added more support for Port Forwarding. [Port Forwarding]
•
Added support for USB/SD Card Files Cleanup. [USB/SD Card Files Cleanup]
•
Added support for A Key Dial-up FXO. [A key Dial-up FXO]
•
Added support for ACIM Detect Option for FXO. [INTERFACE SETTINGS]
•
Added support for some special character on the file name of FW. [Upgrading Via Local Upload]
•
Added more search criteria of CDR. [CDR]
•
Added support of "Allow outgoing calls if registration failure" for register trunks. [Allow outgoing calls if
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registration failure]
•
Added support for music on hold playback from webGUI. [MUSIC ON HOLD]
•
Added support to enable delete recording files for user privilege. [Consumer]
•
Added support disk Inode usage in "Storage Usage" page. [Storage Usage]
•
Added support foe Ring Group/Call Queue/IVR Display Option for Caller ID. [Replace Display Name |
Replace Caller ID | Replace Caller ID]
•
Added support for compatibility between backup package from UCM61xx and UCM62xx.
[Backup/Restore]
•
Added support to Detect talking users in conference. [CONFERENCE ]
•
Added Support of Mini Bar for PMS. [Mini Bar]
Firmware Version 1.0.11.27
•
Added ability to sort extension status on Web GUI.
•
Added one click enable / disable feature code. [Feature Codes]
•
Added Uruguay time zone support. [Auto time updating]
•
Added distinctive ring tone support. [Configure Call Queue] [Configure IVR] [Create New SIP Extension]
•
Added special character support for SFTP client account. [Data Sync]
•
Added destination directory support for data sync. [Data Sync]
•
Added ring group music on hold. [Configure Ring Group]
•
Added CDR multi-email / time condition support. [CDR]
•
Added blacklist anonymous call block. [Blacklist Configurations]
•
Added ability to sort selected extension in Eventlist. [Event List]
•
Added Banned User list for Web GUI login attempts. [Login Settings]
•
Added Email template support. [Email Templates]
•
Added outbound route country restriction.
•
Added external disk usage alert option. [Alert Events List]
•
Added range IP input support for dynamic defense white list. [Dynamic Defense]
•
Added blacklist support for Fail2ban. [Fail2ban]
•
Added ability to reboot device from zero config page. [Discovery]
•
Added GXP1628B template for zero config. [Model Update]
•
Added PIN group support. [PIN Groups]
•
Added PMS support. [PMS]
•
Added call queue custom prompt support. [Configure Call Queue]
•
Added call queue retry time support. [Configure Call Queue]
•
Added Support for DHCP Client List. [DHCP Client List]
Firmware Version 1.0.0.7
•
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.
----------------------------------------------------------------------------------------------------------------------------- --------------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.
---------------------------------------------------------------------------------------------------------------------------------- ----------
This document is subject to change without notice. The latest electronic version of this user manual is available
for download here:
http://www.grandstream.com/support
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any
purpose without the express written permission of Grandstream Networks, Inc. is not permitted.
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PRODUCT OVERVIEW
Technical Specifications
The following table resumes all the technical specifications including the protocols / standards supported, voice
codecs, telephony features, languages and upgrade/provisioning settings for UCM6200 series.
Table 1: Technical Specifications
Interfaces
Analog Telephone FXS
Ports
PSTN Line FXO Ports
Network Interfaces
2 ports (both with lifeline capability in case of power outage)
•
UCM6202: 2 ports
•
UCM6204: 4 ports
•
UCM6208: 8 ports
•
UCM6202/6204/6208: Dual Gigabit RJ45 ports with integrated PoE Plus
(IEEE 802.3at-2009)
NAT Router
Yes
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
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Network Protocols
Disconnect Methods
TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH,
HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS, LADP
Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current
Disconnect, Busy Tone
Security
Media
SRTP, TLS, HTTPS, SSH
Physical
Universal Power Supply
Dimensions
Environmental
Mounting
Weight
•
Output: 12VDC, 1.5A
•
Input: 100-240VAC, 50-60Hz
•
UCM6202/6204: 226mm (L) x 155mm (W) x 34.5mm (H)
•
UCM6208: 440mm (L) x 185mm (W) x 44mm (H)
•
Operating: 32 - 104oF / 0 - 40oC, 10-90% (non-condensing)
•
Storage: 14 - 140oF / -10 - 60oC
•
UCM6202/6204: Wall mount and Desktop
•
UCM6208: Rack mount and Desktop
•
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
English/Simplified Chinese/Traditional Chinese/Spanish/French/
Portuguese/German/Russian/Italian/Polish/Czech for Web GUI; Customizable
Multi-language Support
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
Multiple configurable call queues, automatic call distribution (ACD) based on
Call Center
agent skills/availability busy level, in-queue announcement
Customizable Auto
Attendant
Up to 5 layers of IVR (Interactive Voice Response)
•
UCM6202:
Concurrent audio calls up to 30, concurrent WebRTC calls up to 25.
•
Maximum Call Capacity
UCM6204:
Concurrent audio calls up to 45, concurrent WebRTC calls up to 35.
•
UCM6208:
Concurrent audio calls up to 100, concurrent WebRTC calls up to 50.
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Or up to 66% performance if calls are SRTP encrypted
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SIP Devices
Conference Rooms
•
UCM6202/6204 up to 500 registered SIP endpoints.
•
UCM6208 up to 800 registered SIP endpoints.
•
UCM6202/6204: Up to 3 password-protected conference rooms allowing up
to 25 simultaneous PSTN or IP participants
•
UCM6208: Up to 6 password-protected conference rooms allowing up to 32
simultaneous PSTN or IP participants
Call Features
Compliance
Call park, call forward, call transfer, DND, ring/hunt group, paging/intercom and
etc
•
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)
----------------------------------------------------------------------------------------------------------------------------- ---------------------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: UCM6200 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 WAN 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.
6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax)
to the FXS ports.
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8 x FXO Port
USB Port
SD Card Slot
LED Indicators
Navigation Keys
LCD
2 x FXS Port
Figure 5: UCM6208 Front View
LAN Port
DC 12 V
Reset
Wan Port
Port
Wan
Ground
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.
------------------------------------------------------------------------------------------------------------------ --------------------------------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.
The following table summarizes the layout of the LCD menu of UCM.
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Table 3: LCD Menu Options
View Events
Device Info
Network Info
Network Menu
•
Critical Events
•
Other Events
•
Hardware: Hardware version number
•
Software: Software version number
•
P/N: Part number
•
WAN MAC: WAN side MAC address
•
LAN MAC: LAN side MAC address
•
Uptime: System up time
•
WAN Mode: DHCP, Static IP, or PPPoE
•
WAN IP: IP address
•
WAN Subnet Mask
•
LAN IP: IP address
•
LAN Subnet Mask
•
WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE
•
Static Route Reset: Click to reset the static route setting
•
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
Select "All On" "All Off" or "Blinking" and check LED status.
Factory Menu
•
RTC Test Patterns
Select "2022-02-22 22:22" or "2011-01-11 11:11" to start the RTC (RealTime 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.
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Web Info
SSH Switch
•
Protocol: Web access protocol. HTTP or HTTPS. By default, it's HTTPS
•
Port: Web access port number. By default, it's 8089
•
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
Solid: Connected
SD
Flashing: Data Transferring
FXS (Phone/Fax)
OFF: Not Connected
FXO (Telco Line)
Table 5: UCM6208 LED Indicators
LED
NETWORK
LED Status
Solid: Connected
OFF: Not Connected
ACT
USB
SD
Phone (FXS)
Line (FXO)
Solid: Connected
Flashing: Data Transferring
OFF: Not Connected
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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.
Figure 7: UCM6202 Web GUI Login Page
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.
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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 default administrator username “admin” and password.
Note: Units manufactured starting January 2017 have a unique random password printed on the sticker located
on the back of the unit. It is highly recommended to change the default password after login for the first time.
Older units have default password “admin”. See below pictures for the location of the default random password:
Figure 8: Default Random Password
----------------------------------------------------------------------------------------------------------------------------- ---------------------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 set under the UCM6200 Web GUISystem SettingsHTTP
Server.
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Setup Wizard
When the user logs in the UCM6200 Web GUI 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 9: UCM6200 Setup Wizard
Users can disable the Extension Range during configuration on the Extensions.
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.
Web GUI Configurations
There are eight main sections in the Web GUI for users to view the PBX status, configure and manage the PBX.
•
System Status: Displays PBX status, System Status, Active calls and network status.
•
Extension/Trunk: To configure extensions, trunks and manage inbound/outbound call routes.
•
Call Features: To Configure call features such as ring groups, call queues, IVR ...Etc
•
PBX Settings: Extension ranges, SIP Settings, interfaces settings and MOH.
•
System Settings: To configure user management, network settings, firewall settings, change password,
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LDAP Server, HTTP Server, Email Settings, Time Settings, NTP server and login timeout.
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•
Maintenance: To perform firmware upgrade, backup configurations, cleaner setup, reset/reboot, syslog
setup and troubleshooting.
•
CDR: View call statistics and download CDR and call recordings.
•
Value-added Features: Zero Config, enable CTI server, CRM, PMS WebRTC.
Web GUI Languages
Currently the UCM6200 series Web GUI supports English, Simplified Chinese, Traditional Chinese,
Spanish, French, Portuguese, Russian, Italian, Polish, German 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 10: UCM6200 Web GUI Language
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
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requires reboot to take effect, a prompted message will pop up for you to reboot the device.
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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 Extension/TrunkExtensions.
2. Click on
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
“System Settings” tab on UCM6200 Web GUI. System settings include User Management, Network Settings,
Firewall/Security Settings, Change Password, LDAP server, HTTP server, Email settings, Time Settings, NTP
Server, Recordings Storage and Login Timeout settings.
Network Settings
After successfully connecting the UCM6200 to the network for the first time, users could login the Web GUI and
go to System SettingsNetwork Settings to configure the network parameters for the device.
•
UCM6200 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System SettingsNetwork Settings page to configure LAN settings, WAN settings, 802.1X and Port
Forwarding.
Basic Settings
Please refer to the following tables for basic network configuration parameters on UCM6202, UCM6204 and
UCM6208 respectively.
Table 6: UCM6200 Network SettingsBasic Settings
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.
Method
•
Switch
WAN port interface will be used for uplink connection. LAN port interface will be used
as room 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.
MTU
Specifies the Maximum Transmission Unit. (By default, its 1500)
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IPv4 Address
Preferred DNS
Enter the preferred DNS server address. If Preferred DNS is used, UCM will try to use it
Server
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
Assign the VLAN tag of the layer 2 QoS packets for WAN port. The default value is 0.
Tag
Layer 2 QoS
802.1p Priority
Assign the priority value of the layer 2 QoS packets for WAN port. The default value is 0.
Value
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
Allow IP
Address To
Default IP
Lease Time
Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.
Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.
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
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Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.
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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
Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0.
Tag
Layer 2 QoS
802.1p Priority
Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0.
Value
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
Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0.
Tag
Layer 2 QoS
802.1p Priority
Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0.
Value
IPv6 Address
WAN (when "Method" is set to "Route")
IP Method
Select Auto or Static. The default setting is Auto
IP Address
Enter the IP address for static IP settings.
IP Prefixlen
Enter the Prefix length for static settings. Default is 64
DNS Server 1
Enter the DNS server 1 address for static settings.
DNS Server 2
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Enter the DNS server 2 address for static settings.
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LAN (when Method is set to "Route")
Select Disable, Auto or DHCPv6.
Disable: the DHCPv6 server is disabled.
DHCP Server
Auto: Stateless address auto configuration using NDP protocol.
DHCPv6: Stateful address auto configuration using DHCPv6 protocol.
DHCP Prefix
Enter DHCP prefix. (Default is 2001:db8:2:2::)
DHCP
Enter the Prefix length for static settings. Default is 64
prefixlen
DNS Server 1
Enter the DNS server 1 address for static settings. Default is (2001:4860:4860::8888 )
DNS Server 2
Enter the DNS server 2 address for static settings. Default is (2001:4860:4860::8844 )
Allow IP
Configure starting IP address assigned by the DHCP prefix and DHCP prefixlen.
Address From
Allow IP
Configure the ending IP address assigned by the DHCP Prefix and DHCP prefixlen.
Address To
Default IP
Configure the lease time (in second) of the IP address.
Lease Time
LAN (when Method is set to "Switch")
IP Method
Select Auto or Static. The default setting is Auto
IP Address
Enter the IP address for static IP settings.
IP Prefixlen
Enter the Prefix length for static settings. Default is 64
DNS Server 1
Enter the DNS server 1 address for static settings.
DNS Server 2
Enter the DNS server 2 address for static settings.
LAN 1 / LAN 2 (when Method is set to "Dual")
Users will need assign the default interface to be LAN 1 (mapped to UCM6200 WAN port)
Default
or LAN 2 (mapped to UCM6200 LAN port) and then configure network settings for LAN
Interface
•
1/LAN 2. The default interface is LAN 2.
IP Method
Select Auto or Static. The default setting is Auto
IP Address
Enter the IP address for static IP settings.
IP Prefixlen
Enter the Prefix length for static settings. Default is 64
DNS Server 1
Enter the DNS server 1 address for static settings.
DNS Server 2
Enter the DNS server 2 address for static settings.
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.
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Figure 11: UCM6202 Network Interface Method: Route
•
Method: Switch
WAN port interface is used for uplink connection; LAN port interface is used as room for PC connection.
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Figure 12: UCM6202 Network Interface Method: Switch
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•
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.
Figure 13: UCM6202 Network Interface Method: Dual
DHCP Client List
This feature can bind MAC to IP addresses on the LAN port when UCM6200 is set to Route mode.
When devices receive IP addresses from LAN port, they will be listed on the webGUI under “System
SettingsNetwork SettingsDHCP Client List” as shown below.
Figure 14: DHCP Client List
User can bind manually a MAC to an IP address by clicking on
pop up.
, the following figure will
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Figure 15: Add MAC Address Bind
User needs to set the device MAC address and the IP that will be bound to it (the IP address needs to be within
the UCM6200 DHCP range).
To bind a batch of listed MAC addresses, user needs to check first the MAC addresses to bind and click on
. A confirmation popup will be shown, click
to bind the addresses.
Figure 16: Batch Add MAC Address Bind
After Clicking “OK” to confirm the binding, the “Bind Status” will change from “Unbind” to “Binding”.
802.1X
IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication
mechanism to device before the device can access Internet or other LAN resources. The 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.
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Figure 17: UCM6200 Using 802.1X as Client
Figure 18: 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
to upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated from the RADIUS
server.
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Table 7: UCM6200 Network Settings802.1X
Select 802.1X mode. The default setting is "Disable". The supported 802.1X mode
are:
802.1X Mode
•
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System
Settings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, etc.
•
Click on
to create a new IPv4 static route or click on
to
create a new IPv6 static route. The configuration parameters are listed in the table below.
•
Once added, users can select
•
Select
to edit the static route.
to delete the static route.
Table 8: UCM6200 Network SettingsStatic Routes
Configure the destination IPv4 address or the destination IPv6 subnet for the
UCM6200 to reach using the static route.
Destination
Example:
IPv4 address - 192.168.66.4
IPv6 subnet - 2001:740:D::1/64
Configure the subnet mask for the above destination address. If left blank, the
Netmask
default value is 255.255.255.255.
Example:
255.255.255.0
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Configure the IPv4 or IPv6 gateway address so that the UCM6200 can reach the
Gateway
destination via this gateway. Gateway address is optional.
Example:
192.168.40.5
or
2001:740:D::1
Specify the network interface on the UCM6200 to reach the destination using the
Interface
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.
Figure 19: 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
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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 20: UCM6204 Static Route Configuration
Port Forwarding
The UCM network interface supports router function which provides users the ability to do port forwarding. If
LAN mode is set to "Route" under Web GUISystem SettingsNetwork SettingsBasic Settings page,
port forwarding is available for configuration.
The port forwarding configuration is under Web GUISystem SettingsNetwork SettingsPort Forwarding
page. Please see related settings in the table below.
Table 9: UCM6200 Network SettingsPort Forwarding
Specify the WAN port number or a range of WAN ports. Unlimited number of ports
can be configured.
Note:
WAN Port
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.
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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
LAN Port
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 GUI access to public side.
•
UCM6200 network mode is set to “Route”.
•
UCM6200 WAN port is connected to uplink switch, with a public IP address configured, e.g. 1.1.1.1.
•
UCM6200 LAN port provides DHCP pool that connects to multiple phone devices in the LAN network
192.168.2.x. The UCM6200 is used as a router, with gateway address 192.168.2.1.
•
There is a GXP2160 connected under the LAN interface network of the UCM6200. It obtains IP address
192.168.2.100 from UCM6200 DHCP pool.
•
On the UCM6200 Web GUISystem SettingsNetwork SettingsPort Forwarding, configure a port
forwarding entry as the figure shows below.
•
Click on
WAN Port: This is the port opened on the WAN side for access purpose.
LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM6200.
LAN Port: This is the port opened on the GXP2160 side for access purpose.
Protocol Type: We select TCP here for Web GUI access using HTTP.
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Figure 21: Create New Port Forwarding
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Figure 22: UCM6200 Port Forwarding Configuration
This will allow users to access the GXP2160 Web GUI from public side, by typing in public IP address (example:
1.1.1.1:8088).
Figure 23: GXP2160 Web Access using UCM6202 port forwarding
OpenVPN
Open VPN settings allow the users to configure UCM6200 to use VPN features, the following table gives details about
the various options in order to configure the UCM as OpenVPN Client.
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Table 10: UCM6200 System SettingsNetwork SettingsOpen VPN
Enable
Enable / Disable the open VPN feature.
Server Address
Configures the hostname/IP and port of the server. For example: 192.168.1.2:22
Server Protocol
Specify the protocol user, user should use the same settings as used on the
server
Use the same setting as used on the server.
Device Mode
Dev TUN: Create a routed IP tunnel.
Dev TAP: Create an Ethernet tunnel.
User
Compress tunnel packets using the LZO algorithm on the VPN link. Don’t enable
Compression
this unless it is also enabled in the server config file.
CA Cert
Upload as SSL/TLS root certificate. This file will be renamed as ‘ca.crt’
automatically.
Client Cert
Upload a client certificate. This file will be renamed as ‘cliend.crt’ automatically.
Client Key
Upload a client private key. This file will be renamed as ‘client.key’ automatically.
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Figure 24: Open VPN feature on the UCM6200
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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.
Figure 25: Register Domain Name on noip.com
2. On Web GUISystem SettingsNetwork SettingsDDNS Settings, enable DDNS service and
configure username, password and host name.
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Figure 26: UCM6200 DDNS Setting
3. Now you can use domain name instead of IP address to connect to the UCM6200 Web GUI.
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Figure 27: Using Domain Name to Connect to UCM6200
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Security Settings
The UCM6200 provides users firewall security 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 GUISystem
SettingsSecurity Settings page.
Static Defense
Under Web GUISystem SettingsSecurity Settings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.
Table 11: UCM6200 FirewallStatic DefenseCurrent Service
Port
Process
Type
Protocol or Service
7777
Asterisk
TCP/IPv4
SIP
389
Slapd
TCP/IPv4
LDAP
2000
Asterisk
TCP/IPv4
SCCP
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.
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Table 12: 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 (UCM6200) interface.
Allows the UCM6200 to handle excessive amounts of SYN packets from one
source and keep the web portal accessible. There are two options available and
only one of these options may be enabled at one time.
•
eth(0)LAN defends against attacks directed to the LAN IP address of the
UCM6200.
SYN-Flood
Defense Enable
•
eth(1)WAN defends against attacks directed to the WAN IP address of the
UCM6200.
SYN Flood Defense will limit the amount of SYN packets accepted by the UCM
from one source to 10 packets per second. Any excess packets from that source
will be discarded.
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
(UCM6200) 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 28: Create New Firewall Rule
Table 13: Firewall Rule Settings
Rule Name
Specify the Firewall rule name to identify the firewall rule.
Select the action for the Firewall to perform.
Action
•
ACCEPT
•
REJECT
•
DROP
Select the traffic type.
•
Type
IN
If selected, users will need specify the network interface "LAN" or "WAN"
(for UCM6200) for the incoming traffic.
•
OUT
Select the service type.
Service
•
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
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Save the change and click on "Apply" button. Then submit the configuration by clicking on "Apply Changes" on
the upper right of the web page. The new rule will be listed at the bottom of the page with sequence number,
rule name, action, protocol, type, source, destination and operation. More operations below:
•
Click on
to edit the rule.
•
Click on
to delete the rule.
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System 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 14: UCM6200 Firewall Dynamic Defense
Dynamic Defense
Enable
Enable dynamic defense. The default setting is disabled.
Configure the dynamic defense periodic time interval (in minutes). If the number
Periodical Time
of TCP connections from a host exceeds the connection threshold within this
Interval
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
Configure the blacklist update time interval (in seconds). The default setting is
Interval
120.
Configure the connection threshold. Once the number of connections from the
Connection
same host reaches the threshold, it will be added into the blacklist. The default
Threshold
setting is 100.
Allowed IPs and ports range, multiple IP addresses and port range.
Dynamic Defense
For example:
Whitelist
192.168.5.100192.168.5.200 1500:2000
The following figure shows a configuration example like this:
•
If a host at IP address 192.168.5.7 initiates more than 20 TCP connections to the UCM6200 within 1 minute,
it will be added into UCM6200 blacklist.
•
This host 192.168.5.7 will be blocked by the UCM6200 for 500 seconds.
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•
Since IP range 192.168.5.100-192.168.5.200 is in whitelist, if a host initiates more than 20 TCP connections
to the UCM6200 within 1 minute, it will not be added into UCM6200 blacklist. It can still establish TCP
connection with the UCM6200.
Figure 29: 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 act to forbid the host for certain period as defined in "Banned Duration". This feature helps prevent SIP brute
force attacks to the PBX system.
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Figure 30: Fail2ban Settings
Table 15: Fail2Ban Settings
Global Settings
Enable Fail2Ban. The default setting is disabled. Please make sure both "Enable
Enable Fail2Ban
Fail2Ban" and "Asterisk Service" are turned on 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
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Max Retry Duration
Within this duration (in seconds), if a host exceeds the max times of retry as
defined in "MaxRetry", the host will be banned. The default setting is 600.
Configure the number of authentication failures during "Max Retry Duration" before
MaxRetry
the host is banned. The default setting is 5.
Configure IP address, CIDR mask or DNS host in the whitelist. Fail2Ban will not
Fail2Ban Whitelist
ban the host with matching address in this list. Up to 5 addresses can be added
into the list.
Local Settings
Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make
Asterisk Service
sure both "Enable Fail2Ban" and "Asterisk Service" are turned on to use Fail2Ban
for SIP authentication on the UCM6200.
Listening Port
Configure the listening port number for the service. By default, port 5060 will be
Number
used for UDP and TCP, and port 5061 will be used for TLS.
Configure the number of authentication failures during "Max Retry Duration" before
MaxRetry
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".
Login Attack Defense
Enables defense against excessive login attacks to the UCM’s web GUI.
The default setting is disabled.
Listening Port
This is the Web GUI listening port number which is configured under System
Number
SettingsHTTP ServerPort. The default is 8089.
MaxRetry
When the number of failed login attempts from an IP address exceeds the
MaxRetry number, that IP address will be banned from accessing the Web GUI.
Blacklist
Black List
Users will be able to view the IPs that have been blocked by UCM.
SSH Access
SSH switch now is available via Web GUI and LCD. User can enable or disable SSH access directly from Web
GUI or LCD screen. For web SSH access, please log in UCM6200 web interface and go to Web GUISystem
SettingsSecurity Settings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 31: SSH Access
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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 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 another 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.
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Figure 32: 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 33: Default LDAP Phonebook DN
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Figure 34: 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Extension/TrunkExtensions first. The default LDAP phonebook will then be updated automatically.
Figure 35: LDAP ServerLDAP Phonebook
•
Add new phonebook
A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under "LDAP
Phonebook" section.
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Figure 36: 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 37: Edit LDAP Phonebook
•
Import phonebook from your computer to LDAP server
Click on “Import Phonebook” and a dialog will prompt as shown in the figure below.
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Figure 38: Import Phonebook
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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 39: 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 UCM6200.
Figure 40: LDAP Phonebook After Import
As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in LDAP
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•
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 41: 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
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 configuration for UCM6200 acting as a LDAP client.
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Figure 42: 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 GXP2170 to successfully use the
LDAP server as configured in Figure 32: LDAP Server Configurations.
The users can tune these settings to match with the paramters of their own LDAP server deployed on their
environement.
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Figure 43: GXP2170 LDAP Phonebook Configuration
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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System SettingsHTTP
Server.
Table 16: HTTP Server Settings
Enable or disable redirect from port 80. On the PBX, the default access
Redirect From Port 80
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".
Select HTTP or HTTPS. The default setting is "HTTPS". This is also the
Protocol Type
protocol used for zero config when the end point device downloads the
config file from the UCM6200.
Specify port number to access the HTTP server. The default port number
Port
is 8089.
Enable IP whitelist
If enabled, only the IP address on the permitted IP list will be allowed to
access the UCM’s web GUI.
Add an IP address to the list of allowed IPs to access UCM’s web GUI.
Permitted IP(s)
Ex: 192.168.6.233 / 255.255.255.255
TLS Private Key
Upload private key for the built-in http server
TLS Cert
Upload certificate for the built-in http server and override the existing one.
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.
Time settings
Auto time updating
The current system time on the UCM6200 can be found under Web GUISystem StatusDashboardPBX
Status.
To configure the UCM6200 to update time automatically, go to Web GUISystem SettingsTime
SettingsAuto Time Updating.
----------------------------------------------------------------------------------------------------------------------------- ---------------------Note:
The configurations under Web GUISettingsTime SettingsAuto Time 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 17: Time Auto Updating
Specify the URL or IP address of the NTP server for the UCM6200 to
Remote NTP Server
synchronize
the
date
and
time.
The
default
NTP
server
is
ntp.ipvideotalk.com.
If set to "Yes", the UCM6200 can get provisioned for Time Zone from
Enable DHCP Option 2
DHCP Option 2 in the local server automatically. The default setting is
"Yes".
If set to "Yes", the UCM6200 can get provisioned for NTP Server from
Enable DHCP Option 42
DHCP Option 42 in the local server automatically. This will override the
manually configured NTP Server. The default setting is "Yes".
Select the proper time zone option so the UCM6200 can display correct
time accordingly.
Time Zone
If "Self-Defined Tome Zone" is selected, please specify the time zone
parameters in "Self-Defined Time Zone" field as described in below option.
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
Self-Defined Time Zone
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 GUISystem SettingsTime SettingsSet Time
Manually. The format is YYYY-MM-DD HH:MI:SS.
Figure 44: Set Time Manually
----------------------------------------------------------------------------------------------------------------------------- ---------------------Note:
Manually setup time will take effect immediately after saving and applying change in the Web GUI. If users would
like to reboot the 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Auto Time Updating page are unchecked or set to empty. Otherwise, time auto updating settings in
this page will take effect after reboot.
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NTP Server
The 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System SettingsTime
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.
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
GUISystem SettingsTime SettingsOffice Time. Click on "Create New Office Time" to create an office
time.
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Figure 45: Create New Office Time
Table 18: 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 option 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 46: 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 GUISystem
SettingsTime SettingsHoliday. Click on "Create New Holiday" to create holiday time.
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Figure 47: Create New Holiday
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Table 19: 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
Week
Check this option to show advanced options. If selected, please specify
the days as holiday in one week below.
Select the days as holiday in one week.
Enter holiday "Name" and "Holiday Memo" for the new holiday. Then select "Month" and "Day". The system
administrator can also define days in one week as advanced options. Once done, click on "Save" and then
"Apply Change" for the holiday to take effect. The holiday will be listed in the web page as the figure shows
below.
Figure 48: 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.
----------------------------------------------------------------------------------------------------------------------------- --------------Note:
For more details on how to use office time and holiday, please refer to the link below:
http://www.grandstream.com/sites/default/files/Resources/office_time_and_holiday_on_ucm6xxx.pdf
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Email
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) etc. The configuration parameters can be accessed via Web GUISystem
SettingsEmail SettingsEmail Settings.
Table 20: Email Settings
Enable or disable TLS during transferring/submitting your Email to
TLS Enable
another SMTP server. The default setting is "Yes".
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
Type
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.
Specify the domain name to be used in the Email when using type
Domain
"MTA".
Server
Specify the SMTP server when using type "Client".
Username is required when using type "Client". Normally it's the Email
Username
address.
Password to login for the above Username (Email address) is required
Password
when using type "Client".
Display Name
Specify the display name in the FROM header in the Email.
Specify the sender's Email address.
Sender
For example:
pbx@example.mycompany.com.
The following figure shows a sample Email setting on the UCM6200, assuming the Email is using 192.168.6.202
as the SMTP server.
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Figure 49: 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.
Email Templates
The Email templates on the UCM6200 can be used for email notification the configuration parameters can be
accessed via Web GUISettingsEmail SettingsEmail Templates.
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Figure 50: Email Templates
To configure the email template, click the
button under Options column, and edit the template as desired.
Figure 51: Conference Schedule Template
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-
Users have the ability to preview mail sample by clicking on
-
Click on
-
Finally, users can click on
.
in order to restore the default email template.
to upload a custom picture to the email template to display their
own logo in the sent mails for example
Email Send Log
Under UCM Web GUISystem SettingsEmail SettingsEmail Send Log, users could search, filter and
check whether the Email is sent out successfully or not. This page will also display the corresponding error
message if the Email is not sent out successfully.
Figure 52: Email Send log
Table 21: Email Log
Field
Description
Start Time
Enter the start time for filter
End Time
Enter the end time for filter
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Receivers
Enter the email recipient, while searching for multiple recipients, please
separate then with comma and no spaces.
Send Result
Enter the status of the send result to filter with
Return Code
Enter the email code to filter with
Email Send Module
Select the email module to filter with from the drop-down list, which
contains:
All Modules
Extension
Voicemail
Conference Schedule
User Password
Alert Events
CDR
Fax
Test
Email logs will be shown on bottom of the “Email Send Log” page, as shown on the following figure.
Figure 53: Email Logs
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PBX SettingsRecordings Storage page and select whether to store the recording files
in USB Disk, SD card or locally on the UCM6200.
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Figure 54: 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.
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 55: Recordings Storage Prompt Information
Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to be
copied over.
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Figure 56: 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.
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Call FeaturesConferenceGoogle Service
SettingsGoogle Service Settings.
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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.
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.
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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.
Figure 60: Google ServiceOAuth2.0 Login
6. Now UCM6200 is connected with Google Service.
You can also configure the Status update, which refresh automatically your Google Calendar with the configured
time (m). Note: Zero means disable.
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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
GUIValue-added Features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
Route mode need to be set to use this feature. When the phone restarts (by default DHCP Option 66 is
turned on), it will send out a DHCP DISCOVER request. The UCM6200 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
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Value-added FeaturesZero ConfigZero Config
Settings, fill in the auto provision information.
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Figure 63: Auto Provision Settings
Table 22: Auto Provision Settings
Enable Zero Config
Enable or disable the zero config feature on the PBX. The default setting
is enabled.
By default, this is disabled. If disabled, when SIP device boots up, the
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
Enable Automatic
Configuration Assignment
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
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Segment" to the device. The default setting is disabled.
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Click on the link "Zero Config Extension Segment" to specify the extension
Zero Config Extension
Segment
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 GUIPBX SettingsGeneral SettingsGeneral
pageExtension Preference section: "Auto Provision Extensions".
If enabled, the extension list will be sent out to the device after receiving
Enable Pick Extension
the device's request. This feature is for the GXP series phones that support
selecting extension to be provisioned via phone's LCD. The default setting
is disabled.
Click on the link "Pick Extension Segment" to specify the extension list to
be sent to the device. The default range is 4000 to 4999. Pick Extension
Pick Extension Segment
Segment range can be defined in Web GUIPBX SettingsGeneral
Settings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.
This feature allows the UCM to provision devices in different subnets other
than UCM network.
Enter subnets IP addresses to allow devices within these subnets to be
provisioned. The syntax is <IP>/<CIDR>.
Subnet Whitelist
Examples:
10.0.0.1/8
192.168.6.0/24
Note: Only private IP ranges (10.0.0.0 | 172.16.0.0 | 192.168.0.0) are
supported.
Please make sure an extension is manually assigned to the phone or "Automatically Assign Extension" is
enabled during provisioning. After the configuration on the 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
Grandstream endpoints are automatically discovered after bootup. Users could also manually discover device
by specifying the IP address or scanning the entire LAN network. Three methods are supported to scan the
devices.
•
PING
•
ARP
•
SIP Message (NOTIFY)
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Click on "Auto Discover" under Web GUIValue-added Features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Value-added FeaturesZero ConfigAuto Provisioning Settings.
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 /Reboot /Access
Device WebGUI) are displayed in the list.
Figure 65: Discovered Devices
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Uploading Devices List
Besides the built-in discovery method on the UCM, users could prepare a list of devices on .CSV file and upload it by
clicking on the button
, after which a success message prompt should be displayed.
Users need to make sure that the CSV file respects the format as shown on the following figure and that the entered
information is correct (valid IP address, valid MAC address, device model ..etc), otherwise the UCM will reject the file
and the operation will fail:
Figure 66: Device list - CSV file sample
Managing discovered devices:
•
Sorting: Press
or
to sort per MAC Address, IP Address, Version, Vendor, Model or Create
Config columns from lower to higher or higher to lower respectively.
•
Filter: Select a filter
▪ All: Display all discovered devices.
to display corresponding results.
▪
Scan Results: Display only manually discovered devices. [Discovery]
▪
IP Address: Enter device IP and press Search button.
▪
MAC Address: Enter device MAC and press Search button.
▪
Model: Enter a model name and press Search button. Example: GXP2130.
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Figure 67: Managing Discovered Devices
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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 GUIValue-added FeaturesZero ConfigGlobal Policy
•
Web GUIValue-added FeaturesZero ConfigGlobal Templates.
•
Global Templates configuration has higher priority to Global Policy configuration.
Global Policy can be accessed in Web GUIValue-added Features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:
•
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.
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Figure 68: Global Policy Categories
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The following tables list the Global Policy configuration parameters for the SIP end device.
Table 23: 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.
Configure the time display in 12-hour or 24-hour format on the SIP end device’s
Time Format
LCD.
Configure the URL or IP address of the NTP server. The SIP end device may
NTP Server
obtain the date and time from the server.
Time Zone
Configure the time zone used on the SIP end device.
Table 24: Global Policy ParametersPhone Settings
Default Call Settings
Configure the default dial plan rule. For syntax and examples, please refer to
Dial Plan
user manual of the SIP devices to be provisioned for more details.
When enabled, “Do Not Disturb”, “Call Forward” and other call features can be
Enable Call Features
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.
If set to "Yes", the phone will automatically turn on the speaker phone to answer
Auto Answer by Call-
incoming calls after a short reminding beep, based on the SIP Call-Info header
info
sent from the server/proxy.
The default setting is “Enabled”.
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
NAT Traversal
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.
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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”. Both parties will negotiate to use which port to
allow SIP through.
The default setting is "Keep-alive".
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
Use Random Port
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
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]]];
Call Progress Tones
•
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
Select “Default Mode” or “Toggle Headset/Speaker” for the Headset key.
HEADSET Key Mode
Please refer to user manual of the SIP devices to be provisioned for more
details.
Table 25: Global Policy ParametersContact List
LDAP Phonebook
Select "Manual" or "PBX" as the LDAP configuration source.
•
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
GUISystem 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.
This is the location in the directory where the search is requested to begin. EX:
Base DN
•
dc=grandstream, dc=com
•
ou=Boston, dc=grandstream, dc=com
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Configure the bind "Username" for querying LDAP servers. The field can be
User Name
left blank if the LDAP server allows anonymous binds.
Configure the bind "Password" for querying LDAP servers. The field can be left
Password
blank if the LDAP server allows anonymous binds.
Configure the filter used for number lookups. Please refer to user manual for
Number Filter
more details.
Configure the filter used for name lookups. Please refer to user manual for
Name Filter
more details.
Select the protocol version for the phone to send the bind requests. The default
Version
value is 3.
Specify the "name" attributes of each record which are returned in the LDAP
search result. Example:
Name Attribute
•
gn
•
cn sn description
Specify the "number" attributes of each record which are returned in the LDAP
Number Attribute
search result. Example:
•
telephoneNumber
•
telephoneNumber Mobile
Configure the entry information to be shown on phone's LCD. Up to 3 fields
Display Name
can be displayed. Example:
•
%cn %sn %telephoneNumber
Specify the maximum number of results to be returned by the LDAP server.
Max Hits
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.
Specify whether the searching result is sorted or not.
Sort Results
The default setting is No.
Configure to enable LDAP number searching when receiving calls. The default
Incoming Calls
setting is No.
Configure to enable LDAP number searching when making calls. The default
Outgoing Calls
setting is No.
Lookup Display Name
Configures the display name when LDAP looks up the name for incoming call
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or outgoing call. It must be a subset of the LDAP Name Attributes.
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XML Phonebook
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
Phonebook XML 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
Configure the phonebook download interval (in Minute). If set to 0, automatic
Interval
download will be disabled. Valid range is 5 to 720.
Remove manually-edited
If set to "Yes", when XML phonebook is downloaded, the entries added
entries on download
manually will be automatically removed.
Table 26: Global Policy ParametersMaintenance
Upgrade and Provision
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”.
Firmware Source
•
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
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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.
When URL is selected as firmware source, configure upgrade via TFTP, HTTP
Upgrade via
or HTTPS.
When URL is selected as firmware source, configure the firmware upgrading
Server Path
server path.
When URL is selected as firmware source, configure the firmware file prefix. If
File Prefix
configured, only the firmware with the matching encrypted prefix will be
downloaded and flashed into the phone, if URL is selected as firmware source.
When URL is selected as firmware source, configure the firmware file postfix.
File Postfix
If configured, only the configuration file with the matching encrypted postfix will
be downloaded and flashed into the phone.
Allow DHCP Option
If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can be
43/66
redirected.
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.
•
By week
Once selected, specify the day of the week to check HTTP/TFTP server
for firmware upgrades or configuration files changes.
Automatic Upgrade
•
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.
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Web Server Port
Configure the port for web access. The valid range is 1 to 65535.
Security
Enable Telnet/SSH access for the SIP end device. If the SIP end device
Disable Telnet/SSH
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 27: Global Policy ParametersNetwork Settings
Basic Settings
Configure how the SIP end device shall obtain the IP address. DHCP or PPPoE
can be selected.
•
DHCP
Once selected, users can specify the Host Name (option 12) of the SIP end
IP Address
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
Define the Layer 3 QoS parameter. This value is used for IP Precedence, Diff-
Layer 3 QoS
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
Assign the priority value of the Layer 2 QoS packets. Valid range is 0-7.
Configure the IP address or Domain name of the STUN server. Only non-
STUN Server
symmetric NAT routers work with STUN.
Keep Alive Interval
Specify how often the phone will send a blank UDP packet to the SIP server to
keep the "ping hole" on the NAT router to open. Valid range is 10-160.
Table 28: 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
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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.
Check this option if the SIP end device shall use 800 x 400 resolution for the
LCD screen wallpaper.
Screen Resolution
800 x 400
•
Source
Configure the location where wallpapers are stored.
•
File
If "URL" is selected as source, specify the URL of wallpaper file. If "Local
UCM Server" is selected as source, click to upload wallpaper file to UCM.
Check this option if the SIP end device shall use 480 x 272 resolution for the
LCD screen wallpaper.
•
Screen Resolution
480 x 272
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.
Check this option if the SIP end device supports 320 x 240 resolution for the
LCD screen wallpaper.
•
Screen Resolution
Configure the location where wallpapers are stored.
•
320 x 240
Source
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Value-added Features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.
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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 29: 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 69: Edit Global Template
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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
template. On the right side of each option, users can click on
to remove this option from the
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 GUIValue-added Features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.
•
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 Value-added Features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 30: Create New Model Template
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Select a model to apply this template. The supported Grandstream models
Model
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.
Select to assign this model template as the default model template. The value
Default Model Template
of the option in default model template will be overridden if other selected
model template has a different value for the same option.
Active
•
Click on
Check this option to enable the model template.
to edit the model template.
The editing window for model template is shown in the following figure. In the “Options” field, enter the option
name key word, the option that contains the key word will be listed. User could then select the option 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.
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Figure 70: Edit Model Template
•
Click on Save when done. The model template will be displayed on Web GUIValue-added
Features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 GUIValue-added
Features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”
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•
Click on
to download the template.
•
Click on
to upgrade the model template. Users will see this icon available if the device model has
template updated in the UCM6200.
Figure 71: 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 GUI. Please contact Grandstream customer support if the model
package is needed for manual uploading.
Figure 72: Upload Model Template Manually
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Device Configuration
On Web GUI, page Value-added Features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 73: Create New Device
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Manage Devices
The device manually created or discovered from Auto Discover will be listed in the Web GUIValue-added
FeaturesZero ConfigZero Config page. Users can see the devices with their MAC address, IP address,
vendor, model etc.
Figure 74: Manage Devices
•
Click on
to access the Web GUI of the phone.
•
Click on
to edit the device configuration.
A new dialog will be displayed for the users to configure “Basic” settings and “Advanced” settings. “Basic”
settings have the same configurations as displayed when manually creating a new device, i.e., account, line
key and MPK settings; “Advanced” settings allow users to configure more details in a five-level structure.
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Figure 75: Edit Device
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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 GUIValue-added
FeaturesZero ConfigGlobal Policy will be applied here. Clicking on “Modify Global Policy” to
redirect to page Value-added FeaturesZero ConfigGlobal Policy.
(2) Global Templates
Select a global template to be used for the device and click on
to add. Multiple global templates
can be selected and users can arrange the priority by adjusting orders via
and
. All the selected
global templates will take effect. If the same option exists on multiple selected global templates, the
value in the template with higher priority will override the one in the template with lower priority. Click on
to remove the global template from the selected list.
(3) Default Model Template
Default Model Template will be applied to the devices of this model. Default model template can be
configured in model template under Web GUIValue-added FeaturesZero ConfigModel
Templates page. Please see default model template option in [Table 30: 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 76: 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 77: 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 78: 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 79: 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 GUIValue-added Features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 80: 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Value-added FeaturesZero ConfigZero Config Settings, select “Enable Zero
Config”.
2. Go to Web GUIValue-added FeaturesZero ConfigGlobal Policy, configure Date Format, Time
Format and Firmware Source as follows.
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Figure 81: Zero Config Sample - Global Policy
3. Go to Web GUIValue-added Features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Value-added FeaturesZero ConfigModel Templates, create another model template
“Spanish Support Template” for GXP2140. Add option “Language” and set it to “Español”.
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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 GUIValue-added FeaturesZero
ConfigZero Config.
6. On Web GUIValue-added Features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.
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 82: 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 83: Zero Config Sample - Device Preview 2
Select “Spanish Support Template” in “Model Template”. The preview of the device settings is displayed on the
right side and we can see the language is set to “Español” since Model Template has the higher priority for the
option “Language”, which overrides the value configured in default model template.
9. For the GXV3275 used by the customer support supervisor, select an available extension for account 1 on
“Basic” settings and click on “Save”. Users can see the preview of the device configuration in “Advanced”
settings. There is no model template configured for GXV3275.
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Figure 84: Zero Config Sample - Device Preview 3
10. Click on “Apply Changes” to apply saved changes.
11. On the Web GUIValue-added FeaturesZero ConfigZero Config page, click on
to trigger the device to download config file from UCM6200.
to send NOTIFY
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Extension/TrunkExtensions. Click on "Add" and a new
dialog window will show for users to fill in the extension information.
Figure 85: Create New Device
Extension options are divided into four categories:
•
Basic Settings
•
Media
•
Features
•
Specific Time
•
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Select first which type of extension: SIP Extension, IAX Extension or FXS Extension. The configuration
parameters are as follows.
Table 31: SIP Extension Configuration ParametersBasic Settings
General
Extension
The extension number associated with the user.
Configure the CallerID Number that would be applied for outbound calls from this
user.
CallerID Number
Note:
The ability to manipulate your outbound Caller ID may be limited by your VoIP
provider.
Assign permission level to the user. The available permissions are "Internal", "Local",
"National" and "International" from the lowest level to the highest level. The default
setting is "Internal".
Permission
Note:
Users need to have the same level as or higher level than an outbound rule's privilege
to make outbound calls using this rule.
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.
Configure the authentication ID for the user. If not configured, the extension number
Auth ID
will be used for authentication.
Enable Voicemail
Enable voicemail for the user. The default setting is "Yes".
Configure voicemail password (digits only) for the user to access the voicemail box.
Voicemail
A random numeric password is automatically generated. It is recommended to use
Password
the random generated password for security purpose.
Skip Voicemail
When user dials voicemail code, the password verification IVR is skipped. If enabled,
Password
this would allow one-button voicemail access. By default, this option is disabled.
Verification
Allows users to send voicemail recording as .wav file attachment to specified email
Send Voicemail to
addresses. Default setting is “No”.
Email
Note: When set to “Default”, the global settings in Call Features  Voicemail 
Voicemail Email Settings will be used.
Whether to keep the local voicemail recording after sending them. If set to “Default”,
Keep Voicemail
the global settings will be used.
after Emailing
Note: When set to “Default”, the global settings in Call Features  Voicemail 
Voicemail Email Settings will be used.
Enable Keep-alive
Keep-alive
Frequency
If enabled, empty SDP packet will be sent to the SIP server periodically to keep the
NAT port open. The default setting is "Yes".
Configure the Keep-alive interval (in seconds) to check if the host is up. The default
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If selected, this extension will be disabled on the UCM6200.
Disable This
Note:
Extension
The disabled extension still exists on the PBX but can’t be used on the end device.
User Settings
Configure the first name of the user. The first name can contain characters, letters,
First Name
digits and _.
Configure the last name of the user. The last name can contain characters, letters,
Last Name
digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email address.
Configure the password for user portal access. A random numeric password is
User Password
automatically generated. It is recommended to use the randomly generated password
for security purpose.
Select the voice prompt language to be used for this extension. The default setting is
"Default" which is the selected voice prompt language under Web GUIPBX
SettingsVoice PromptLanguage Settings. The dropdown list shows all the
Language
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 SettingsVoice PromptLanguage Settings.
Concurrent
The maximum endpoints which can be registered into this extension. For security
Registrations
concerns, the default value is 1.
Configure the phone number for the extension, user can type the related star code for
Mobile Phone
phone number followed by the extension number to call directly this number.
Number
Example: user can type *881000 to call the mobile number associated with extension
1000.
Table 32: SIP Extension Configuration ParametersMedia
SIP Settings
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
NAT
related to NAT configuration or Firewall's support of SIP and RTP ports. The default
setting is enabled.
By default, the UCM6200 will route the media steams from SIP endpoints through
Can Direct Media
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".
Select DTMF mode for the user to send DTMF. The default setting is "RFC2833". If
DTMF Mode
"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
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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
TEL URI
“TO” header in the SIP request to indicate the E.164 number. If set to “Enable”, “Tel”
will be used instead of “SIP” in the SIP request.
When present in an INVITE request, the alert-Info header field specifies and
Alert-Info
alternative ring tone to the UAS.
Enable T.38
Enable or disable T.38 UDPTL support.
UDPTL
SRTP
Enable SRTP for the call. The default setting is disabled.
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
Fax Mode
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 GUICall
FeaturesFax/T.38.
•
Yes – Allow Fax to Email for this extension. Faxes will be sent to the user’s
email address configured in the extension’s Basic Settings.
Fax To Email
•
No – Do not send any faxes to the user’s email address configured in the
extension’s Basic Settings.
This option controls how the extension can be used on devices within different types
of network. The default setting is "Allow All".
•
Allow All
Device in any network can register this extension.
•
Strategy
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.
Select audio and video codec for the extension. The available codecs are: PCMU,
Codec Preference
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 33: SIP Extension Configuration ParametersFeatures
Call Transfer
Select which presence status to set for the extension and configure call forward
Presence Status
conditions for each status. Six possible options are possible: “Available”, “Away”,
“Chat”, “Custom”, “DND” and “Unavailable”. More details at [PRESENCE].
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Enable and configure the Call Forward Unconditional target number. Available options
for target number are:
Call Forward
Unconditional
•
“None”: Call forward deactivated.
•
“Extension”: Select an extension from dropdown list as CFU target.
•
“Custom Number”: Enter a customer number as target. For example: *97.
•
“Voicemail”: Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension.
•
“Ring Group”: Select a ring group from dropdown list as CFU target.
•
“Queues”: Select a queue from dropdown list as CFU target.
•
“Voicemail Group”: Select a voicemail group from dropdown list as CFU target.
The default setting is “None”.
Select time condition for Call Forward Unconditional. CFU takes effect only during the
selected time condition. The available time conditions are “Office Time”, “Out of Office
Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.
Note:
CFU Time
•
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
Condition
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Configure the Call Forward No Answer target number. Available options for target
number are:
Call Forward No
Answer
•
“None”: Call forward deactivated.
•
“Extension”: Select an extension from dropdown list as CFN target.
•
“Custom Number”: Enter a customer number as target. For example: *97.
•
“Voicemail”: Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension.
•
“Ring Group”: Select a ring group from dropdown list as CFN target.
•
“Queues”: Select a queue from dropdown list as CFN target.
•
“Voicemail Group”: Select a voicemail group from dropdown list as CFN target.
The default setting is “None”.
Select time condition for Call Forward No Answer. The available time conditions are
CFN Time
Condition
“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or
Holiday” and “Specific”.
Note:
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•
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Configure the Call Forward Busy target number. Available options for target number
are:
Call Forward Busy
•
“None”: Call forward deactivated.
•
“Extension”: Select an extension from dropdown list as CFB target.
•
“Custom Number”: Enter a customer number as target. For example: *97.
•
“Voicemail”: Select an extension from dropdown list. Incoming calls will be
forwarded to voicemail of selected extension.
•
“Ring Group”: Select a ring group from dropdown list as CFB target.
•
“Queues”: Select a queue from dropdown list as CFB target.
•
“Voicemail Group”: Select a voicemail group from dropdown list as CFB target.
The default setting is “None”.
Select time condition for Call Forward Busy. The available time conditions are “Office
Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”
and “Specific”.
Note:
CFB Time
•
Condition
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Do Not Disturb
If enabled the extension will ignore all incoming calls
Select time condition for Do Not Disturb. The available time conditions are “Office
Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”
and “Specific”.
Note:
DND Time
•
Condition
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
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If DND is enabled, all calls to this extension will be rejected except the numbers listed
on this list.
DND Whitelist
Note: The maximum number on the Whitelist is 10.
If call forward is enabled, all calls to this extension will be forwarded except the calls
coming from the specified numbers on this list.
FWD Whitelist
Note: The Maximum number on the whitelist is 10.
CC Settings
If enabled, UCM6200 will automatically alert this extension when a called party is
Enable CC
available, given that a previous call to that party failed for some reason. By default,
it’s disabled.
Two modes for Call Completion are supported:
CC Mode
•
Normal: This extension is used as ordinary extension.
•
For Trunk: This extension is registered from a PBX.
The default setting is “Normal”.
Configure the maximum number of CCSS agents which may be allocated for this
CC Max Agents
channel. In other words, this number serves as the maximum number of CC requests
this channel can make. The minimum value is 1.
Configure the maximum number of monitor structures which may be created for this
CC Max Monitors
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
Enable this option to have an external number ring simultaneously along with the
Ring
extension. If a register trunk is used for outbound, the register number will be used to
Simultaneously
be displayed for the external number as caller ID number.
External Number
Time Condition for
Ring
Use callee DOD on
Simultaneously
which will be ignored.
Ring the external number simultaneously along with the extension on the basis of this
time condition.
Simultaneously
FWD or Ring
Set the external number to be rang simultaneously. ‘-’ is the connection character
Use the DOD number when calls are being diverted/forwarded to external
destinations or when ring simultaneous is configured.
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Monitor privilege control
Allowed to
Add members from “Available Extensions” to “Selected Extensions” so that the
call-barging
selected extensions can spy on the used extension using feature code.
Seamless transfer privilege control
Any extensions on the UCM can perform seamless transfer. When using Pickup Incall
Allowed to
seamless transfer
feature, only extensions available on the “Selected Extensions” list can perform
seamless transfer to the edited extension.
Other Settings
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
Ring Timeout
global
ring
timeout
setting
under
Web
GUIPBX
SettingsVoice
PromptCustom 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.
Enable automatic recording for the calls using this extension. The default setting is
Auto Record
disabled. The recordings can be accessed under Web GUICDRRecording Files.
•
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
Skip Trunk Auth
calls during the selected time condition.
•
If set to “No”, users will be asked to enter the password when making outbound
calls.
Time Condition for
If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which users can
Skip Trunk Auth
skip entering password when making outbound calls.
Dial Trunk
Configure personal password when making outbound calls via trunk.
Password
Support Hot-
If enabled, SIP Password will accept only alphabet characters and digits. Auth ID will
Desking Mode
be changed to the same as Extension.
Enable LDAP
If enabled, the extension will be added to LDAP Phonebook PBX list.
Enable WebRTC
Enable registration and call from WebRTC.
Support
Specify which Music On Hold class to suggest to the bridged channel when putting
Music On Hold
them on hold.
Call Duration Limit
Maximum Call
Duration
The maximum duration of call-blocking.
The maximum call duration (in seconds). The default value 0 means no limit.
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Custom
Call-info
for Auto Answer
If enabled, when a call is sent to this extension from UCM, the SIP INVITE message
will contain a Call-info header indicating auto answer.
Table 34: 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 like SIP but also has its own characteristic. For more
information, please refer to RFC 5465.
To manually create new IAX user, go to Web GUIExtension/TrunkExtensions. Click on "Add" and a new
dialog window will show for users which need to make sure first to select the extension type to be IAX Extension
before proceeding to fill in the extension information. The configuration parameters are as follows.
Table 35: IAX Extension Configuration ParametersBasic Settings
General
Extension
The extension number associated with the user.
Configure the CallerID Number that would be applied for outbound calls from this
CallerID Number
user.
Note: The ability to manipulate your outbound Caller ID may be limited by your VoIP
provider.
Assign permission level to the user. The available permissions are "Internal", "Local",
"National" and "International" from the lowest level to the highest level. The default
Permission
setting is "Internal".
Note: Users need to have the same level as or higher level than an outbound rule's
privilege to make outbound calls using this rule.
SIP/IAX Password
Enable Voicemail
Voicemail
Password
Skip Voicemail
Password
Verification
Configure the password for the user. A random secure password will be automatically
generated. It is recommended to use this password for security purpose.
Enable voicemail for the user. The default setting is "Yes".
Configure voicemail password (digits only) for the user to access the voicemail box.
A random numeric password is automatically generated. It is recommended to use
the random generated password for security purpose.
When user dials voicemail code, the password verification IVR is skipped. If enabled,
this would allow one-button voicemail access. By default, this option is disabled.
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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
Last Name
Email Address
Configure the first name of the user. The first name can contain characters, letters,
digits and _.
Configure the last name of the user. The last name can contain characters, letters,
digits and _.
Fill in the Email address for the user. Voicemail will be sent to this Email address.
Configure the password for user portal access. A random numeric password is
User Password
automatically generated. It is recommended to use the randomly generated password
for security purpose.
Select the voice prompt language to be used for this extension. The default setting is
"Default" which is the selected voice prompt language under Web GUIPBX
Language
SettingsVoice PromptLanguage Settings. 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 SettingsVoice PromptLanguage Settings.
Table 36: IAX Extension Configuration ParametersMedia
IAX Settings
Max Number of
Calls
Require Call
Token
SRTP
Configure the maximum number of calls allowed for each remote IP address.
Configure to enable/disable requiring call token. If set to "Auto", it might lock out users
who depend on backward compatibility when peer authentication credentials are
shared between physical endpoints. The default setting is "Yes".
Enable SRTP for the call. The default setting is disabled.
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
Fax Mode
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 GUICall
FeaturesFax/T.38.
This option controls how the extension can be used on devices within different types
of network.
Strategy
•
Allow All
Device in any network can register this extension.
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Local Subnet Only
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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".
Select audio and video codec for the extension. The available codecs are: PCMU,
Codec Preference
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 37: IAX Extension Configuration ParametersFeatures
Call Transfer
Configure the Call Forward Unconditional target number. If not configured, the Call
Call Forward
Forward Unconditional feature is deactivated.
Unconditional
The default setting is deactivated.
Select time condition for Call Forward Unconditional. CFU takes effect only during the
selected time condition. The available time conditions are “Office Time”, “Out of Office
Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.
Note:
•
CFU Time
Condition
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Call Forward No
Configure the Call Forward No Answer target number. If not configured, the Call
Answer
Forward No Answer feature is deactivated. The default setting is deactivated.
Select time condition for Call Forward No Answer. The available time conditions are
“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or
Holiday” and “Specific”.
Note:
•
CFN Time
Condition
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Call Forward Busy
Configure the Call Forward Busy target number. If not configured, the Call Forward
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Busy feature is deactivated. The default setting is deactivated.
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Select time condition for Call Forward Busy. The available time conditions are “Office
Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”
and “Specific”.
Note:
•
CFB Time
Condition
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Ring Simultaneously
Enable this option to have an external number ring simultaneously along with the
Ring
extension. If a register trunk is used for outbound, the register number will be used to
Simultaneously
be displayed for the external number as caller ID number.
External Number
Time Condition for
Ring
Set the external number to be rang simultaneously. ‘-’ is the connection character
which will be ignored.
Ring the external number simultaneously along with the extension on the basis of this
time condition.
Simultaneously
Other Settings
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
Ring Timeout
global
ring
timeout
setting
under
Web
GUIPBX
SettingsVoice
PromptCustom 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.
Enable automatic recording for the calls using this extension. The default setting is
Auto Record
disabled. The recording files can be accessed under Web GUICDRRecording
Files.
•
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
Skip Trunk Auth
calls during the selected time condition.
•
If set to “No”, users will be asked to enter the password when making outbound
calls.
Time Condition for
If “Skip Trunk Auth” is set to “By Time”, select a time condition during which users can
Skip Trunk Auth
skip entering password when making outbound calls.
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Dial Trunk
Configure personal password when making outbound calls via trunk.
Password
Enable LDAP
If enabled, the extension will be added to LDAP Phonebook PBX lists.
Configure the Music On Hold class to suggest to the bridged channel when putting
Music On Hold
them on hold.
Call Duration Limit
The maximum duration of call-blocking.
Table 38: 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Extension/TrunkExtensions. Click on "Add" and a new
dialog window will show for users which need to make sure first to select the extension type to be FXS Extension
before proceeding to fill in the extension information. The configuration parameters are as follows.
Table 39: 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.
Configure the CallerID Number that would be applied for outbound calls from this
Caller ID Number
user. Note:
The ability to manipulate your outbound Caller ID may be limited by your VoIP
provider.
Assign permission level to the user. The available permissions are "Internal", "Local",
"National" and "International" from the lowest level to the highest level. The default
setting is "Internal".
Permission
Note:
Users need to have the same level as or higher level than an outbound rule's privilege
to make outbound calls using this rule.
Enable Voicemail
Voicemail
Password
Enable voicemail for the user. The default setting is "Yes".
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.
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Skip Voicemail
When user dials voicemail code, the password verification IVR is skipped. If enabled,
Password
this would allow one-button voicemail access. By default, this option is disabled.
Verification
If selected, this extension will be disabled on the UCM6200.
Disable This
Note:
Extension
The disabled extension still exists on the PBX but can’t be used on the end device.
User Settings
Configure the first name of the user. The first name can contain characters, letters,
First Name
digits and _.
Configure the last name of the user. The last name can contain characters, letters,
Last Name
digits and _.
Email Address
Fill in the Email address for the user. Voicemail will be sent to this Email address.
Configure the password for user portal access. A random numeric password is
User Password
automatically generated. It is recommended to use the randomly generated password
for security purpose.
Select the voice prompt language to be used for this extension. The default setting is
"Default" which is the selected voice prompt language under Web GUIPBX
SettingsVoice PromptLanguage Settings. The dropdown list shows all the
Language
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 SettingsVoice PromptLanguage Settings.
Table 40: FXS Extension Configuration ParametersMedia
Analog Settings
Call Waiting
Configure to enable/disable call waiting feature. The default setting is "No".
User ‘#’ as SEND
RX Gain
TX Gain
If configured, the # key can be used as SNED key after dialing the number on the
analog phone. The default setting is "Yes".
Configure the RX gain for the receiving channel of analog FXS port. The valid range
is -30dB to +6dB. The default setting is 0.
Configure the TX gain for the transmitting channel of analog FXS port. The valid range
is -30dB to +6dB. The default setting is 0.
Configure the minimum period of time (in milliseconds) that the hook-flash must
MIN RX Flash
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.
Configure the maximum period of time (in milliseconds) that the hook-flash must
MAX RX Flash
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.
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If enabled, a polarity reversal will be marked as received when an outgoing call is
Enable Polarity
answered by the remote party. For some countries, a polarity reversal is used for
Reversal
signaling the disconnection of a phone line and the call will be considered as Hangup
on a polarity reversal. The default setting is "Yes".
Specify "ON", "OFF" or a value (the power of 2) from 32 to 1024 as the number of
taps of cancellation.
Echo Cancellation
Note: When configuring the number of taps, the number 256 is not translated into
256ms of echo cancellation. Instead, 256 taps mean 256/8 = 32 ms. The default
setting is "ON", which is 128 taps.
3-Way Calling
Send CallerID
After
Configure to enable/disable 3-way calling feature on the user. The default setting is
enabled.
Configure the number of rings before sending CID. Default setting is 1.
For FXS extension, there are three options available in Fax Mode. The default setting
is “None”.
•
None: Disable Fax.
•
Fax 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.
Fax Mode
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 GUICall
Features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. only for FXS ports.
Table 41: FXS Extension Configuration ParametersFeatures
Call Transfer
Call Forward
Configure the Call Forward Unconditional target number. If not configured, the Call
Unconditional
Forward Unconditional feature is deactivated. The default setting is deactivated.
Select time condition for Call Forward Unconditional. CFU takes effect only during the
selected time condition. The available time conditions are “Office Time”, “Out of Office
Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.
Note:
CFU Time
•
Condition
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
Call Forward No
Answer
Configure the Call Forward No Answer target number. If not configured, the Call
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Forward No Answer feature is deactivated. The default setting is deactivated.
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Select time condition for Call Forward No Answer. The available time conditions are
“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or
Holiday” and “Specific”.
Note:
•
CFN Time
Condition
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System 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.
Select time condition for Call Forward Busy. The available time conditions are “Office
Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”
and “Specific”.
Note:
CFB Time
•
Condition
“Specific” has higher priority to “Office Times” if there is a conflict in terms of time
period.
•
Specific time can be configured on the bottom of the extension configuration
dialog. Scroll down the add Time Condition for specific time.
•
Office Time and Holiday could be configured on page System SettingsTime
SettingsOffice Time/Holiday page.
CC Settings
If enabled, UCM6200 will automatically alert this extension when a called party is
Enable CC
available, given that a previous call to that party failed for some reason.
Ring Simultaneously
Enable this option to have an external number ring simultaneously along with the
Ring
extension. If a register trunk is used for outbound, the register number will be used to
Simultaneously
be displayed for the external number as caller ID number.
External Number
Time Condition for
Ring
Simultaneously
Set the external number to be rang simultaneously. ‘-’ is the connection character
which will be ignored.
Ring the external number simultaneously along with the extension on the basis of this
time condition.
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Hotline
If enabled, hotline dialing plan will be activated, a pre-configured number will be used
Enable Hotline
according to the selected Hotline Type.
Hotline Number
Configure the Hotline Number
Configure the Hotline Type:
•
Immediate Hotline:
When the phone is off-hook, UCM6200 will immediately dial the preset number
Hotline Type
•
Delay Hotline:
When the phone is off-hook, if there is no dialing within 5 seconds, UCM6200 will
dial the preset number.
Other Settings
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
Ring Timeout
global
ring
timeout
setting
under
Web
GUIPBX
SettingsVoice
PromptCustom 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.
Enable automatic recording for the calls using this extension. The default setting is
Auto Record
disabled. The recording files can be accessed under Web GUICDRRecording
Files.
•
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
Skip Trunk Auth
calls during the selected time condition.
•
If set to “No”, users will be asked to enter the password when making outbound
calls.
Time Condition for
If “Skip Trunk Auth” is set to “By Time”, select a time condition during which users can
Skip Trunk Auth
skip entering password when making outbound calls.
Dial Trunk
Configure personal password when making outbound calls via trunk.
Password
If enabled, this extension will be added to LDAP Phonebook PBX list; if disabled, this
Enable LDAP
extension will be skipped when creating LDAP Phonebook.
Select which Music On Hold class to suggest to extension when putting the active call
Music On Hold
on hold.
Call Duration Limit
Configure the maximum duration of call-blocking.
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Table 42: 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
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Extension/TrunkExtensions, click on "Add” and select extension type as SIP extension,
and “add method” as Batch.
Table 43: 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.
Extension Interval
Specify the interval between extensions as preferred when creating a batch of
extension.
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
Permission
setting is "Internal".
Note: Users need to have the same level as or higher level than an outbound rule's
privilege to make outbound calls from this rule.
Enable Voicemail
Enable WebRTC
Enable Voicemail for the user. The default setting is "Yes".
Enable WebRTC support.
Support
Configure the SIP/IAX password for the users. Three options are available to create
password for the batch of extensions.
•
SIP/IAX Password
User Random Password.
A random secure password will be automatically generated. It is recommended
to use this password for security purpose.
•
Use Extension as Password.
•
Enter a password to be used on all the extensions in the batch.
Configure Voicemail password (digits only) for the users.
•
User Random Password.
Voicemail
A random password in digits will be automatically generated. It is recommended
Password
to use this password for security purpose.
•
Use Extension as Password.
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•
Enter a password to be used on all the extensions in the batch.
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Configure CallerID Number when adding Batch Extensions.
•
CallerID Number
Use Extension as Number
Users can choose to use the extension number as CallerID
•
Use as Number
Users can choose to set a specific number instead of using the extension number.
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PBX
SettingsVoice
PromptCustom Prompt: General Preference. The valid range is between 5
Ring Timeout
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.
Enable automatic recording for the calls using this extension. The default setting is
Auto Record
disabled. The recording files can be accessed under Web GUICDRRecording
Files.
Skip Voicemail
When user dials voicemail code, the password verification IVR is skipped. If enabled,
Password
this would allow one-button voicemail access. By default, this option is disabled.
Verification
Select which Music On Hold class to suggest to extensions when putting them on
Music On Hold
hold.
If enabled, the batch added extensions will be added to LDAP Phonebook PBX list; if
Enable LDAP
disabled, the batch added extensions will be skipped when creating LDAP
Phonebook.
Enable WebRTC
If enabled, extensions will be able to login to user portal and use Web RTC features.
Support
Call Duration Limit
Configure the maximum duration of call-blocking.
SIP Settings
Use NAT when the PBX is on a public IP communicating with devices hidden behind
NAT
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.
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By default, the PBX will route the media steams from SIP endpoints through itself. If
Can Direct Media
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".
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
DTMF Mode
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
Configure the number of seconds for the host to be up for Keep-alive. The default
Frequency
setting is 60 seconds.
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
TEL URI
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
The maximum endpoints which can be registered into this extension. For security
Registrations
concerns, the default value is 1.
Other Settings
Enable SRTP for the call.
SRTP
The default setting is "No".
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
Fax Mode
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 GUICall
FeaturesFax/T.38.
This option controls how the extension can be used on devices within different types
Strategy
of network. The default setting is "Allow All".
•
Allow All
Device in any network can register this extension.
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•
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.
Enable T.38
Enable or disable T.38 UDPTL Support.
UDPTL
If enable “All”, users do not need to enter password when making an outbound call. If
Skip Trunk Auth
enable “Follow Me”, the user can dial out via follow me without password.
Select audio and video codec for the extension. The available codecs are: PCMU,
Codec Preference
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Extension/TrunkExtensions, click on “Add”, then select extension type as IAX Extension
and the add method to be Batch.
Table 44: 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.
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
Permission
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”.
Configure the SIP/IAX password for the users. Three options are available to create
password for the batch of extensions.
•
SIP/IAX Password
User Random Password.
A random secure password will be automatically generated. It is recommended
to use this password for security purpose.
•
Use Extension as Password.
•
Enter a password to be used on all the extensions in the batch.
Configure Voicemail password (digits only) for the users.
•
User Random Password.
Voicemail
A random password in digits will be automatically generated. It is recommended
Password
to use this password for security purpose.
•
Use Extension as Password.
•
Enter a password to be used on all the extensions in the batch.
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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PBX
SettingsVoice
PromptCustom Prompt: General Preference. The valid range is between 5
Ring Timeout
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.
Enable automatic recording for the calls using this extension. The default setting is
Auto Record
disabled. The recording files can be accessed under Web GUICDRRecording
Files.
Skip Voicemail
When user dials voicemail code, the password verification IVR is skipped. If enabled,
Password
this would allow one-button voicemail access. By default, this option is disabled.
Verification
Select which Music On Hold class to suggest to extensions when putting them on
Music On Hold
hold.
If enabled, the batch added extensions will be added to LDAP Phonebook PBX list;
Enable LDAP
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 address.
Configure to enable/disable requiring call token. If set to "Auto", it might lock out
Require Call Token
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".
Select Fax Mode for this user. The default setting is “None”.
Fax Mode
•
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 GUICall
FeaturesFax/T.38.
Strategy
This option controls how the extension can be used on devices within different types
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•
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".
If enable “All”, users do not need to enter password when making an outbound call.
Skip Trunk Auth
If enable “Follow Me”, the call can dial out via follow me without password.
Select audio and video codec for the extension. The available codecs are: PCMU,
Codec Preference
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Extension/Trunk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.
Figure 86: Manage Extensions
•
Status
Users can see the following icon for each extension to indicate the SIP status.
Green:
Blue:
Idle
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•
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Value-added FeaturesZero ConfigZero Config Settings.
•
Delete single extension
Click on
to delete the extension. Or select the checkbox of the extension and then click on "Delete
Selected Extensions".
•
Modify selected extensions
Select the checkbox for the extension(s). Then click on "Edit" to edit the extensions in a batch.
•
Delete selected extensions
Select the checkbox for the extension(s). Then click on "Delete " to delete the extension(s).
Export Extensions
The extensions configured on the 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 87: Export Extensions
The exported csv file can serve as a template for users to fill in desired extension information to be imported to
the UCM6200.
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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 88: Import Extensions
4. Select the option in "On Duplicate Extension" to define how the duplicate extension(s) in the imported csv
file should be treated by the PBX.
•
Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension
information as previously configured without change.
•
Delete and Recreate: The current extension previously configured will be deleted and the duplicate
extension in the csv file will be loaded to the PBX.
•
Update Information: The current extension previously configured in the PBX will be kept. However, if
the duplicate extension in the csv file has different configuration for any options, it will override the
configuration for those options in the extension.
5. Click on “Choose file to upload” to select csv file from local directory in the PC.
6. Click on "Apply Changes" to apply the imported file on the UCM6200.
Example of file to import:
Figure 89: Import File
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Table 45: SIP extensions Imported File Example
Field
Supported values
Extension
Digits
Technology
SIP/SIP(WebRTC)
Enable Voicemail
yes/no
CallerID Number
Digits
SIP/IAX Password
Alphanumeric characters
Voicemail Password
Digits
Skip Voicemail Password Verification
yes/no
Ring Timeout
Empty/ 3 to 600 (in second)
SRTP
yes/no
Fax Mode
None/Fax Detection
Strategy
Allow All/Local Subnet Only/A Specific IP Address
Local Subnet 1
IP address/Mask
Local Subnet 2
IP address/Mask
Local Subnet 3
IP address/Mask
Local Subnet 4
IP address/Mask
Local Subnet 5
IP address/Mask
Local Subnet 6
IP address/Mask
Local Subnet 7
IP address/Mask
Local Subnet 8
IP address/Mask
Local Subnet 9
IP address/Mask
Local Subnet 10
IP address/Mask
Specific IP Address
IP address
Skip Trunk Auth
yes/no/bytime
Codec Preference
PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus
Permission
Internal/Local/National/International
NAT
yes/no
DTMF Mode
RFC2833/info/inband/auto
Insecure
Port
Enable Keep-alive
Yes/no
Keep-alive Frequency
Value from 1-3600
AuthID
Alphanumeric value without special characters
TEL URI
Disabled/user=phone/enabled
Call Forward Busy
Digits
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Call Forward No Answer
Digits
Call Forward Unconditional
Digits
Support Hot-Desking Mode
Yes/no
Dial Trunk Password
Digits
Disable This Extension
Yes/no
CFU Time Condition
All time/Office time/out of office time/holiday/out of
holiday/out of office time or holiday/specific time
CFN Time Condition
All time/Office time/out of office time/holiday/out of
holiday/out of office time or holiday/specific time
CFB Time Condition
All time/Office time/out of office time/holiday/out of
holiday/out of office time or holiday/specific time
Music On Hold
Default/ringbacktone_default
CC Agent Policy
If CC is disabled use: never
If CC is set to normal use: generic
If CC is set to trunk use: native
CC Monitor Policy
Generic/never
CCBS Available Timer
3600/4800
CCNR Available Timer
3600/7200
CC Offer Timer
60/120
CC Max Agents
Value from 1-999
CC Max Monitors
Value from 1-999
Ring simultaneously
Yes/no
External Number
Digits
Time Condition for Ring Simultaneously
All time/Office time/out of office time/holiday/out of
holiday/out of office time or holiday/specific time
Time Condition for Skip Trunk Auth
All time/Office time/out of office time/holiday/out of
holiday/out of office time or holiday/specific time
Enable LDAP
Yes/no
Enable T.38 UDPTL
Yes/no
Max Contacts
Values from 1-10
Enable WebRTC
Yes/no
Alert-Info
None/Ring 1/Ring2/Ring3/Ring 4/Ring 5/Ring 6/Ring 7/ Ring
8/Ring 9/Ring 10/bellcore-dr1/bellcore-dr2/ bellcore-dr3/
bellcore-dr4/ bellcore-dr5/custom
Do Not Disturb
Yes/no
DND Time Condition
All time/Office time/out of office time/holiday/out of
holiday/out of office time or holiday/specific time
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Custom Auto answer
Yes/no
Do Not Disturb Whitelist
Empty/digits
User Password
Alphanumeric characters.
First Name
Alphanumeric without special characters.
Last Name
Alphanumeric without special characters.
Email Address
Email address
Language
Default/en/zh
Phone Number
Digits
Call-Barging Monitor
Extensions allowed to call barging
Seamless Transfer Members
Extensions allowed to seamless transfer
Table 46: IAX extensions Imported File Example
Field
Supported values
Extension
Digits
Technology
IAX
Enable Voicemail
yes/no
CallerID Number
Digits
SIP/IAX Password
Alphanumeric characters
Voicemail Password
Digits
Skip Voicemail Password Verification
yes/no
Ring Timeout
Empty/ 3 to 600 (in second)
SRTP
yes/no
Fax Mode
None/Fax Detection
Strategy
Allow All/Local Subnet Only/A Specific IP Address
Local Subnet 1
IP address/Mask
Local Subnet 2
IP address/Mask
Local Subnet 3
IP address/Mask
Local Subnet 4
IP address/Mask
Local Subnet 5
IP address/Mask
Local Subnet 6
IP address/Mask
Local Subnet 7
IP address/Mask
Local Subnet 8
IP address/Mask
Local Subnet 9
IP address/Mask
Local Subnet 10
IP address/Mask
Specific IP Address
IP address
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Skip Trunk Auth
yes/no/bytime
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Codec Preference
PCMU,PCMA,GSM,G.726,G.722,G.729,H.264,ILBC,AAL2G.726-32,ADPCM,G.723,H.263,H.263p,vp8,opus
Permission
Internal/Local/National/International
NAT
yes/no
Call Forward Busy
Digits
Call Forward No Answer
Digits
Call Forward Unconditional
Digits
Require Call Token
Yes/no/auto
Max Number of Calls
Values from 1-512
Dial Trunk Password
Digits
Disable This Extension
Yes/no
CFU Time Condition
All time/Office time/out of office time/holiday/out of holiday/out
CFN Time Condition
of office time or holiday/specific time
CFB Time Condition
All time/Office time/out of office time/holiday/out of holiday/out
of office time or holiday/specific time
Music On Hold
Default/ringbacktone_default
Ring simultaneously
Yes/no
External Number
Digits
Time Condition for Ring Simultaneously
All time/Office time/out of office time/holiday/out of holiday/out
of office time or holiday/specific time
Time Condition for Skip Trunk Auth
All time/Office time/out of office time/holiday/out of holiday/out
of office time or holiday/specific time
Enable LDAP
Yes/no
Limit Max time (s)
empty
Do Not Disturb
Yes/no
DND Time Condition
All time/Office time/out of office time/holiday/out of holiday/out
of office time or holiday/specific time
Do Not Disturb Whitelist
Empty/digits
User Password
Alphanumeric characters.
First Name
Alphanumeric without special characters.
Last Name
Alphanumeric without special characters.
Email Address
Email address
Language
Default/en/zh
Phone Number
Digits
Call-Barging Monitor
Extensions allowed to call barging
Seamless Transfer Members
Extensions allowed to seamless transfer
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Table 47: FXS Extensions Imported File Example
Field
Supported values
Extension
Digits
Technology
FXS
Analog Station
FXS1/FXS2
Enable Voicemail
yes/no
CallerID Number
Digits
Voicemail Password
Digits
Skip Voicemail Password Verification
yes/no
Ring Timeout
Empty/ 3 to 600 (in second)
Auto Record
yes/no
Fax Mode
None/Fax Detection
Skip Trunk Auth
Yes/no/bytime
Permission
Internal/Local/National/International
Call Forward Busy
Digits
Call Forward No Answer
Digits
Call Forward Unconditional
Digits
Call Waiting
Yes/no
Use # as SEND
Yes/no
RX Gain
Values from -306
TX Gain
Values from -306
MIN RX Flash
Values from: 30 – 1000
MAX RX Flash
Values from: 40 – 2000
Enable Polarity Reversal
Yes/no
Echo Cancellation
On/Off/32/64/128/256/512/1024
3-Way Calling
Yes/no
Send CallerID After
1/2
Dial Trunk Password
digits
Disable This Extension
Yes/no
CFU Time Condition
All time/Office time/out of office time/holiday/out of holiday/out
of office time or holiday/specific time
CFN Time Condition
All time/Office time/out of office time/holiday/out of holiday/out
of office time or holiday/specific time
CFB Time Condition
All time/Office time/out of office time/holiday/out of holiday/out
of office time or holiday/specific time
Music On Hold
Default/ringbacktone_default
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CC Agent Policy
If CC is disabled use: never
If CC is set to normal use: generic
If CC is set to trunk use: native
CC Monitor Policy
Generic/never
CCBS Available Timer
3600/4800
CCNR Available Timer
3600/7200
CC Offer Timer
60/120
CC Max Agents
Value from 1-999
CC Max Monitors
Value from 1-999
Ring simultaneously
Yes/no
External Number
Digits
Time Condition for Ring Simultaneously
All time/Office time/out of office time/holiday/out of holiday/out
Time Condition for Skip Trunk Auth
of office time or holiday/specific time
Enable LDAP
Yes/no
Enable Hotline
Yes/no
Hotline Type
Immediate hotline/delay hotline
Hotline Number
digits
Limit Max time (s)
empty
Do Not Disturb
Yes/no
DND Time Condition
All time/Office time/out of office time/holiday/out of holiday/out
of office time or holiday/specific time
Do Not Disturb Whitelist
Empty/digits
User Password
Alphanumeric characters.
First Name
Alphanumeric without special characters.
Last Name
Alphanumeric without special characters.
Email Address
Email address
Language
Default/en/zh
Phone Number
Digits
Call-Barging Monitor
Extensions allowed to call barging
Seamless Transfer Members
Extensions allowed to seamless transfer
The CSV file should contain all the above fields, if one of them is missing or empty, the UCM6200 will display
the following error message for missing fields.
Figure 90: Import Error
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E-mail Notification
Once the extensions are created with Email addresses, the PBX administrator can click on button “E-mail
Notification” to send the account registration and configuration information to the user. Please make sure Email
setting under Web GUISystem SettingsEmail Settings is properly configured and tested on the UCM6200
before using “E-mail Notification”.
When click on “E-mail Notification” button, the following message will be prompted in the web page. Click on OK
to confirm sending the account information to all users’ Email addresses.
Figure 91: E-mail Notification - 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 92: Account Registration Information and QR Code
Figure 93: LDAP Client Information and QR Code
Multiple Registrations per Extension
UCM6200 supports multiple registrations per extension so that users can use the same extension on devices in
different locations.
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Figure 94: Multiple Registrations per Extension
This
feature
can
be
enabled
by
configuring
option
“Concurrent
Registrations”
under
Web
GUIExtension/TrunkEdit Extension. The default value is set to 1 for security purpose. Maximum is 10.
Figure 95: 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 which is built with Asterisk 13.
Figure 96: SMS Message Support
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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Extension/TrunkExtension Groups.
•
Click on
•
Click on
to edit the extension group.
•
Click on
to delete the extension group.
to create a new extension group.
Select extensions from the list on the left side to the right side.
Figure 97: Edit Extension Group
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Using Extension Groups
Here is an example where the extension group can be used. Go to Web GUIExtension/TrunkOutbound
Routes and select "Enable Filter on Source Caller ID". Both single extensions and extension groups will show
up for users to select.
Figure 98: Select Extension Group in Outbound Route
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ANALOG TRUNKS
Go to Web GUIExtension/Trunk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 48: Analog Trunk Configuration Parameters
Select the channel for the analog trunk.
Channels
•
UCM6202: 2 channels
•
UCM6204: 4 channels
•
UCM6208: 8 channels
Specify a unique label to identify the trunk when listed in outbound rules, incoming
Trunk Name
rules and etc.
Enable this option to satisfy two primary use cases, which include emulating a simple
SLA Mode
key system and creating shared extensions on a PBX. Enable SLA Mode will disable
polarity reversal.
The barge option specifies whether other stations can join a call in progress on this
Barge Allowed
trunk. If enabled, the other stations can press the line button to join the call.
The default setting is Yes.
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
Hold Access
to “Private”, only the station that places the call on hold can retrieve the call.
The default setting is Yes.
Advanced Options
If enabled, a polarity reversal will be marked as received when an outgoing call is
Enable Polarity
Reversal
answered by the remote party. For some countries, a polarity reversal is used for
signaling the disconnection of a phone line and the call will be considered as “Hangup”
on a polarity reversal.
The default setting is “No”.
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When FXO port answers the call, FXS may send a Polarity Reversal. If this interval is
Polarity on
shorter than the value of “Polarity on Answer Delay”, the Polarity Reversal will be
Answer Delay
ignored. Otherwise, the FXO will Onhook to disconnect the call. The default setting is
600ms.
Current
This is the periodic time (in ms) that the UCM6200 will use to check on a voltage drop
Disconnect
in the line. The default setting is 200. The valid range is 50 to 3000.
Threshold (ms)
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
Ring Timeout
to configure how long it takes before the UCM6200 considers a non-ringing line with
Hangup activity. The default setting is 8000.
Configure the RX gain for the receiving channel of analog FXO port. The valid range
RX Gain
is from -13.5 (dB) to + 12.0 (dB). The default setting is 0.
Configure the TX gain for the transmitting channel of analog FXO port. The valid range
TX Gain
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
Configure the time interval between off-hook and first dialed digit for outbound calls.
Delay(ms)
Enable automatic recording for the calls using this trunk. The default setting is
Auto Record
disabled. The recording files can be accessed under Web GUICDRRecording
Files.
Disable This Trunk
If selected, the trunk will be disabled.
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
DAHDI Out Line
When the call goes out from this analog trunk, it will use the port that is not used
Selection
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
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).
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The default setting is “Ascend” mode.
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Tone Settings
Busy Detection is used to detect far end Hangup or for detecting busy signal. The
Busy Detection
default setting is "Yes".
If "Busy Detection" is enabled, users can specify the number of busy tones to be
played before hanging up. The default setting is 2. Better results might be achieved if
Busy Tone Count
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
Congestion detection is used to detect far end congestion signal. The default setting
Detection
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.
Select the country for tone settings. If "Custom" is selected, users could manually
Tone Country
configure the values for Busy Tone and Congestion Tone. The default setting is
"United States of America (USA)".
Syntax:
f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];
Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000)
Busy Tone
Busy Level Range: (-300, 0)
Cadence Range: [0, 16383].
Select Tone Country "Custom" to manually configure Busy Tone value.
Default value:
f1=480@-50,f2=620@-50,c=500/500
Syntax:
f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];
Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000)
Congestion Tone
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
Click on "Detect" to detect the busy tone, Polarity Reversal and Current Disconnect
PSTN Detection
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".
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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Extension/Trunk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 99: UCM6200 FXO Tone Settings
4. Click on "Detect" to start PSTN detection.
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Figure 100: UCM6200 PSTN Detection
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•
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 101: UCM6200 PSTN Detection: Auto Detect
•
If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.
Figure 102: UCM6200 PSTN Detection: Semi-Auto Detect
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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 49: PSTN Detection for Analog Trunk
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.
Detect Model
•
Semi-auto Detect
Semi-auto detection requires answering or hanging up the call
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.
----------------------------------------------------------------------------------------------------------------------------- ---------------------Note:
•
The PSTN detection process will keep the call up for about 1 minute.
•
If "Semi-auto Detect' is used, please pick up the call only after informed from the Web GUI prompt.
•
Once the detection is successful, the detected parameters "Busy Tone", "Polarity Reversal" and "Current
Disconnect by PSTN" will be filled into the corresponding fields in the analog trunk configuration.
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VOIP TRUNKS
VoIP Trunk Configuration
VoIP trunks can be configured in UCM6200 under Web GUIExtension/Trunk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.
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 50: Create New SIP Trunk
Select the VoIP trunk type.
Type
•
Peer SIP Trunk
•
Register SIP Trunk
Configure a unique label to identify this trunk when listed in outbound rules,
Provider Name
inbound rules etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the trunk.
Keep the CID from the inbound call when dialing out. This setting will override
Keep Original CID
“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”.
Turn on this setting when the PBX is using public IP and communicating with
NAT
devices behind NAT. If there is one-way audio issue, usually it is related to NAT
configuration or SIP/RTP port support on the firewall.
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If checked, the trunk will be disabled.
Disable This Trunk
Note: If a current SIP trunk is disabled, UCM will send UNREGISTER message
(REGISTER message with expires=0) to the SIP provider.
If the trunk has an assigned PSTN telephone number, this field should be set to
"User=Phone". Then a "User=Phone" parameter will be attached to the Request-
TEL URI
Line and TO header in the SIP request to indicate the E.164 number. If set to
"Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting
is disabled.
Configure the Caller ID. This is the number that the trunk will try to use when
making outbound calls. For some providers, it might not be possible to set the
CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine which
CallerID will be used if they exist:
Caller ID
•
The CallerID configured for the extension will be looked up first.
•
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 SettingsGeneral Settings will be used.
Need Registration
Username
Password
Auth ID
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.
Enter the username to register to the trunk from the provider when "Register SIP
Trunk" type is selected.
Enter the password to register to the trunk from the provider when "Register SIP
Trunk" is selected.
Enter the Authentication ID for "Register SIP Trunk" type.
Enable automatic recording for the calls using this trunk (for SIP trunk only). The
Auto Record
default setting is disabled. The recording files can be accessed under Web
GUICDRRecording Files.
Table 51: SIP Register Trunk Configuration Parameters
Basic Settings
Provider Name
Host Name
Configure a unique label to identify this trunk when listed in outbound rules,
inbound rules etc.
Configure the IP address or URL for the VoIP provider’s server of the trunk.
Configure the SIP transport protocol to be used in this trunk. The default setting is
"UDP".
Transport
•
UDP
•
TCP
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•
TLS
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Keep the CID from the inbound call when dialing out. This setting will override
Keep Original CID
“Keep Trunk CID” option. Please make sure that the peer PBX at the other side
supports to match user entry using “username” field from authentication line.
If enabled, the trunk CID will not be overridden by extension’s CID when the
Keep Trunk CID
extension has CID configured. The default setting is “No”.
Turn on this option when the PBX is using public IP and communicating with
NAT
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.
If selected, the trunk will be disabled.
Disable This Trunk
Note:
If a current SIP trunk is disabled, UCM will send UNREGISTER message
(REGISTER message with expires=0) to the SIP provider.
If the trunk has an assigned PSTN telephone number, this field should be set to
"User=Phone". Then a "User=Phone" parameter will be attached to the Request-
TEL URI
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
Allow outgoing calls
if registration failure
CallerID Name
Select whether the trunk needs to register on the external server or not when
"Register SIP Trunk" type is selected. The default setting is No.
If enabled outgoing calls even if the registration to this trunk fail will still be able to
go through.
Note that if we uncheck “Need Registration” option, this option will be ignored.
Configure the new name of the caller when the extension has no CallerID Name
configured.
Configure the actual domain name where the extension comes from. This can be
From Domain
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.
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
From User
trunk) with multiple DIDs.
For example, "1234567" is the From User in From Header:
sip:1234567@trunk.UCM6200.provider.com.
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 when "Register SIP Trunk" is selected.
Auth ID
Enter the Authentication ID for "Register SIP Trunk" type.
Auth Trunk
If enabled, the UCM will send 401 response to the incoming call to authenticate
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Enable automatic recording for the calls using this trunk (for SIP trunk only). The
Auto Record
default setting is disabled. The recording files can be accessed under Web
GUICDRRecording Files.
Advanced Settings
Select audio and video codec for the VoIP trunk. The available codecs are: PCMU,
Codec Preference
PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM,
H.264, H.263, H.263p and VP8.
If enabled, the SIP INVITE message sent to the trunk will contain PPI (P-PreferredIdentity) header. The default setting is “No”.
Send PPI Header
Note:
“Send PPI Header” and “Send PAI Header” cannot be enabled at the same time.
Only one of the two headers can be contained in SIP INVITE message.
•
Default – Include the trunk’s preferred CID (configured in Basic Settings)
in the PPI Header.
PPI Mode
•
Original CID – Include the original CID in the PPI Header.
•
DOD Number – Include the trunk’s DOD number in the PPI Header. If no
DOD number has been set, the trunk’s preferred CID will be used.
If enabled, the SIP INVITE message sent to the trunk will contain PAI (P-AssertedSend PAI Header
Identity) header including configured PAI Header. The default setting is “No”.
Note: “Send PPI Header” and “Send PAI Header” cannot be enabled at the same
time. Only one of the two headers can be contained in the SIP INVITE message.
If “Send PAI Header” is enabled and “PAI Header” is configured as “123456” for
instance, the PAI header in the SIP message sent from the UCM will contain
“123456”. If “Send PAI Header” is enabled and “PAI Header” is configured as
PAI Header
“empty”, the PAI header in the SIP message sent from the UCM will contain the
original CID.
Note:
“Send PAI Header” needs to be enabled to use this feature
Outbound Proxy
Support
Outbound Proxy
Remove OBP from
Route
DID Mode
Select to enable outbound proxy in this trunk. The default setting is "No".
When outbound proxy support is enabled, enter the IP address or URL of the
outbound proxy.
It is used to set if the phone system will remove outbound proxy URI from the route
header. If is set to “Yes”, it will remove the route header from SIP requests. The
default setting is “No”.
Configure where to get the destination ID of an incoming SIP call, from SIP
Request-line or To-header. The default is set to "Request-line".
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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 GUIPBX SettingsSIP
SettingsToS.
DTMF Mode
•
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 Heartbeat
If enabled, the UCM6200 will regularly send SIP OPTIONS to the device to check
Detection
if the device is still online. The default setting is "No".
When "Enable Heartbeat Detection" option is set to "Yes", configure the interval
Heartbeat Frequency
(in seconds) of the SIP OPTIONS message sent to the device to check if the device
is still online. The default setting is 60 seconds.
Maximum Number of
The maximum number of concurrent calls using the trunk. The default settings 0,
Call Lines
which means no limit.
Select Fax mode. The default setting is “None”.
Fax Mode
•
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
GUICall FeaturesFax/T.38.
SRTP
Enable SRTP for the VoIP trunk. The default setting is "No".
CC Settings
If enabled, the system will automatically alert the user when a called party is
Enable CC
available, given that a previous call to that party failed for some reason.
Configure the maximum number of CCSS agents which may be allocated for this
CC Max Agents
channel. In other words, this number serves as the maximum number of CC
requests this channel is allowed to make. The minimum value is 1.
Configure the maximum number of monitor structures which may be created for
CC Max Monitors
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.
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Table 52: SIP Peer Trunk Configuration Parameters
Basic Settings
Configure a unique label to identify this trunk when listed in outbound rules,
Provider Name
inbound rules and etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the trunk.
Configure the SIP transport protocol to be used in this trunk. The default setting is
"UDP".
Transport
•
UDP
•
TCP
•
TLS
Keep the CID from the inbound call when dialing out, this setting will override “Keep
Keep Original CID
Trunk CID” option. Please make sure that the peer PBX at the other side supports
to match user entry using “username” field from authentication line.
If enabled, the trunk CID will not be overridden by extension’s CID when the
Keep Trunk CID
extension has CID configured. The default setting is “No”.
Turn on this option when the PBX is using public IP and communicating with
NAT
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.
If selected, the trunk will be disabled. Note:
Disable This Trunk
If a current SIP trunk is disabled, UCM will send UNREGISTER message
(REGISTER message with expires=0) to the SIP provider.
If the trunk has an assigned PSTN telephone number, this field should be set to
"User=Phone". Then a "User=Phone" parameter will be attached to the Request-
TEL URI
Line and TO header in the SIP request to indicate the E.164 number. If set to
"Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting
is disabled.
Configure the Caller ID. This is the number that the trunk will try to use when
making outbound calls. For some providers, it might not be possible to set the
CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine which
Caller ID
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 SettingsGeneral Settings will be used.
CallerID Name
Configure the name of the caller to be displayed when the extension has no
CallerID Name configured.
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Enable automatic recording for the calls using this trunk (for SIP trunk only). The
Auto Record
default setting is disabled. The recording files can be accessed under Web
GUICDRRecording Files.
Advanced Settings
Select audio and video codec for the VoIP trunk. The available codecs are: PCMU,
Codec Preference
PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, iLBC, ADPCM,
H.264, H.263, H.263p and VP8.
Configure where to get the destination ID of an incoming SIP call, from SIP
DID Mode
Request-line or To-header. The default is set to "Request-line".
Configure the default DTMF mode when sending DTMF on this trunk.
•
Default: The global setting of DTMF mode will be used. The global setting
for DTMF Mode setting is under Web GUIPBX SettingsSIP
SettingsToS.
DTMF Mode
•
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 is used.
Enable Heartbeat
If enabled, the UCM6200 will regularly send SIP OPTIONS to the device to check
Detection
if the device is still online. The default setting is "No".
When "Enable Heartbeat Detection" option is set to "Yes", configure the interval
Heartbeat Frequency
(in seconds) of the SIP OPTIONS message sent to the device to check if the device
is still online. The default setting is 60 seconds.
Maximum Number of
The maximum number of concurrent calls using the trunk. The default settings 0,
Call Lines
which means no limit.
Select Fax mode. The default setting is “None”.
Fax Mode
•
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
GUICall FeaturesFax/T.38.
SRTP
Enable SRTP for the VoIP trunk. The default setting is "No".
If enabled, the local UCM6200 will automatically provide and update the local
Sync LDAP Enable
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".
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This is the password used for LDAP contact file encryption and decryption during
Sync LDAP
the LDAP sync process. The password must be the same on both UCM6200 peers
Password
to ensure successful synchronization.
Sync LDAP Port
Configure the TCP port used LDAP sync feature between two peer UCM6200.
LDAP Outbound Rule
LDAP Dialed Prefix
Specify an outbound rule for LDAP sync feature. The UCM6200 will automatically
modify the remote contacts by adding prefix parsed from this rule.
Specify the prefix for LDAP sync feature. The UCM6200 will automatically modify
the remote contacts by adding this prefix.
CC Settings
If enabled, the system will automatically alert the user when a called party is
Enable CC
available, given that a previous call to that party failed for some reason.
Configure the maximum number of CCSS agents which may be allocated for this
CC Max Agents
channel. In other words, this number serves as the maximum number of CC
requests this channel can make. The minimum value is 1.
Configure the maximum number of monitor structures which may be created for
CC Max Monitors
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 53: Create New IAX Trunk
Select the VoIP trunk type.
Type
•
Peer IAX Trunk
•
Register IAX Trunk
Configure a unique label to identify this trunk when listed in outbound rules,
Provider Name
inbound rules etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the trunk.
If enabled, the trunk CID will not be overridden by extension's CID when the
Keep Trunk CID
extension has CID configured. The default setting is "No".
Enter the username to register to the trunk from the provider when "Register IAX
Username
Trunk" type is selected.
Enter the password to register to the trunk from the provider when "Register IAX
Password
Trunk" type is selected.
Disable This Trunk
If selected, the trunk will be disabled.
Table 54: IAX Register Trunk Configuration Parameters
Basic Settings
Provider Name
Configure a unique label to identify this trunk when listed in outbound rules,
inbound rules etc.
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Host Name
Configure the IP address or URL for the VoIP provider’s server of the trunk.
If enabled, the trunk CID will not be overridden by extension's CID when the
Keep Trunk CID
extension has CID configured. The default setting is "No".
Disable This Trunk
If selected, the trunk will be disabled.
Configure the Caller ID. This is the number that the trunk will try to use when
making outbound calls. For some providers, it might not be possible to set the
CallerID with this option and this option will be ignored. When making outgoing
calls, following rules are used to determine which CallerID will be used if they exist:
Caller ID
•
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 SettingsGeneral Settings will be used.
Configure the name of the caller to be displayed when the extension has no
CallerID Name
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
Select audio and video codec for the VoIP trunk. The available codecs are: PCMU,
Codec Preference
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 Heartbeat
If enabled, the UCM6200 will regularly send SIP OPTIONS to the device to check
Detection
if the device is still online. The default setting is "No".
When "Enable Heartbeat Detection" option is set to "Yes", configure the interval
Heartbeat Frequency
(in seconds) of the SIP OPTIONS message sent to the device to check if the device
is still online. The default setting is 60 seconds.
Maximum Number of
The maximum number of concurrent calls using the trunk. The default settings 0,
Call Lines
which means no limited.
Select Fax mode. The default setting is “None”.
Fax Mode
•
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
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GUICall FeaturesFax/T.38.
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Table 55: IAX Peer Trunk Configuration Parameters
Basic Settings
Configure a unique label to identify this trunk when listed in outbound rules,
Provider Name
inbound rules etc.
Host Name
Configure the IP address or URL for the VoIP provider’s server of the trunk.
If enabled, the trunk CID will not be overridden by extension's CID when the
Keep Trunk CID
extension has CID configured. The default setting is "No".
Disable This Trunk
If selected, the trunk will be disabled.
Configure the Caller ID. This is the number that the trunk will try to use when
making outbound calls. For some providers, it might not be possible to set the
CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine which
CallerID will be used if they exist:
Caller ID
•
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 SettingsGeneral Settings will be used.
Configure the name of the caller to be displayed when the extension has no
CallerID Name
CallerID Name configured.
Advanced Settings
Select audio and video codec for the VoIP trunk. The available codecs are: PCMU,
Codec Preference
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 Heartbeat
If enabled, the UCM6200 will regularly send SIP OPTIONS to the device to check
Detection
if the device is still online. The default setting is "No".
When "Enable Heartbeat Detection" option is set to "Yes", configure the interval
Heartbeat Frequency
(in seconds) of the SIP OPTIONS message sent to the device to check if the device
is still online. The default setting is 60 seconds.
Maximum Number of
The maximum number of concurrent calls using the trunk. The default settings 0,
Call Lines
which means no limited.
Select Fax mode. The default setting is “None”.
Fax Mode
•
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
GUICall Features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.
Now
when a user makes an outbound call their caller ID shows up as the main office number. This poses a problem
as the CEO would like their calls to come from their direct line. This can be accomplished by configuring DOD
for the CEO’s extension.
Steps on how to configure DOD on the UCM6200:
1. To setup DOD go to UCM6200 Web GUIExtension/Trunk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. If extension number need to be appended to the DID number click on “Add Extension”.
6. Select an extension from the "Available Extensions" list. Users have the option of selecting more than one
extension. In this case, Company ABC would select the CEO's extension. After making the selection, click
on the
button to move the extension(s) to the "Selected Extensions" list.
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Figure 103: DOD extension selection
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7. 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 104: 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 like 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Extension/TrunkSLA Station.
Figure 105: SLA Station
•
Click on
•
Click on
to edit the SLA Station. The following table shows the SLA Station configuration parameters.
•
Click on
to delete the SLA Station.
to add a SLA Station.
Table 56: 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.
Select a trunk for this SLA from the Available SLA Trunks list. Click on
Selected SLA Trunks
to arrange the order. If there are multiple trunks selected, when there are
calls on those trunks at the same time, pressing the LINE key on the phone will
pick up the call on the first trunk here.
SLA Station Options
Ring Timeout
Configure the time (in seconds) to ring the station before the call is considered
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unanswered. No timeout is set by default. If set to 0, there will be no timeout.
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Configure the time (in seconds) for delay before ringing the station when a call
Ring Delay
first coming in on the shared line. No delay is set by default. If set to 0, there will
be no delay.
This option defines the competence of the hold action for one particular trunk. If
set to “open”, any station could hold a call on that trunk or resume one held
Hold Access
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 GUIExtension/Trunk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 GUIExtension/TrunkSLA Station
page.
Figure 106: Enable SLA Mode for Analog Trunk
2. Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.
Figure 107: Analog Trunk with SLA Mode Enabled
3. On the UCM6200, go to Web GUIExtension/TrunkSLA Station page, click on “Add”. Please refer to
section [Create/Edit SLA Station] for the configuration parameters. Users can create one or more SLA
stations to monitor the analog trunk. The following figure shows two stations, 1002 and 1005, are configured
to be associated with SLA trunk “fxo1”.
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Figure 108: SLA Example - SLA Station
4. On the SIP phone 1, configure to register UCM6200 extension 1002. Configure the MPK as BLF mode and
the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case.
5. On the SIP phone 2, configure to register 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 109: SLA Example - MPK Configuration
Now the SLA station is ready to use. The following functions can be achieved by this configuration.
•
Making an outbound call from the station/extension, using LINE key
When the extension is in idle state, pressing the line key for this extension on the phone to off hook. Then
dial the station’s extension number, for example, dial 1002 on phone 1 (or dial 1005 on phone 2), to hear the
dial tone. Then the users could dial external number for the outbound call.
•
Making an outbound call from the station/extension, using BLF key
When the extension is in idle state, pressing the MPK and users could dial external numbers directly.
•
Answering call using LINE key
When the station is ringing, pressing the LINE key to answer the incoming call.
•
Barging-in active call using BLF key
When there is an active call between an SLA station and an external number using the SLA trunk, other SLA
stations monitoring the same trunk could join the call by pressing the BLF key if “Barge Allowed” is enabled
for the analog trunk.
•
Hold/UnHold using BLF key
If the external line is previously put on hold by an SLA station, another station that monitors the same SLA
trunk could UnHold the call by pressing the BLF key if “Hold Access” is set to “open” on the analog trunk and
the SLA station.
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CALL ROUTES
Outbound Routes
In the following sections, we will discuss the steps and parameters used to configure and manage outbound
rules in UCM6200, these rules are the regulating points for all external outgoing calls initiated by the UCM
through all types of trunks: SIP, Analog and Digital.
Configuring Outbound Routes
In the 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Extension/TrunkOutbound Routes to add and edit outbound rules.
•
Click on
•
Click on
to edit the outbound route.
•
Click on
to delete the outbound route.
to add a new 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.
Table 57: 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.
•
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.
Pattern
".": Wildcard. Match one or more characters.
"!": Wildcard. Match zero or more characters immediately.
Example: [12345-9] - Any digit from 1 to 9.
Notes:
▪
Multiple patterns can be used. Each pattern should be entered in new line.
▪
Users can add comments to a dial plan by typing “/*” and “*/” before and
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▪
Example:
_X.
_NNXXNXXXXX /* 10-digit long distance */
_818X. /* Any number with leading 818 */
After creating the outbound route, users can choose to enable and disable it. If the
Disable This Route
route is disabled, it will not take effect anymore. However, the route settings will
remain in UCM. Users can enable it again when it’s needed.
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.
Maximum Call
Configure the maximum duration of the call (in seconds). The default setting is 0,
Duration
which means no limit.
Configure the warning time for the call using this outbound route. If set to x
Warning Time
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.
Select privilege level for the outbound rule.
Privilege Level
•
Internal: The lowest level required. All users can use this rule.
•
Local: Users with Local, National, or International level can use this rule.
•
National: Users with National or International level can use this rule.
•
International: The highest level required. Only users with international level
can use this rule.
•
Disable: The default setting is "Disable". If selected, only the matched source
caller ID will be allowed to use this outbound route.
Please be aware of the potential security risks when using "Internal" level, which
means all users can use this outbound rule to dial out from the trunk.
When enabled, users could specify extensions allowed to use this outbound route.
"Privilege Level" is automatically disabled if using "Enable Filter on Source Caller
ID".
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 list. This allows users
to specify arbitrary single extensions available in the PBX.
2. Custom Dynamic Route: define the pattern for the source caller ID. This
allows users to define extension range instead of selecting them one by one.
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•
All patterns are prefixed with the "_".
•
Special characters:
X: Any Digit from 0-9.
Z: Any Digit from 1-9.
N: Any Digit from 2-9.
".": Wildcard. Match one or more characters.
"!": Wildcard. Match zero or more characters immediately.
Example: [12345-9] - Any digit from 1 to 9.
Note: Multiple patterns can be used. Patterns should be separated by
comma “,”. Example: _X. , _NNXXNXXXXX , _818X.
Send This Call Through Trunk
Use Trunk
Select the trunk for this outbound rule.
Allows the user to specify the number of digits that will be stripped from the
beginning of the dialed string before the call is placed via the selected trunk.
Example:
Strip
The users will dial 9 as the first digit of a long-distance calls. However, 9 should
not be sent out via analog lines and the PSTN line. In this case, 1 digit should be
stripped before the call is placed.
Specify the digits to be prepended before the call is placed via the trunk. Those
Prepend
digits will be prepended after the dialing number is stripped.
Use 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.
Failover Trunk
UCM62XX support up to 10 failover trunks.
Example:
The user's primary trunk is a VoIP trunk and the user would like to use the PSTN
when the VoIP trunk is not available. The PSTN trunk can be configured as the
failover trunk of the VoIP trunk.
Allows the user to specify the number of digits that will be stripped from the
beginning of the dialed string before the call is placed via the selected trunk.
Strip
Example:
The users will dial 9 as the first digit of a long-distance calls. However, 9 should
not be sent out via analog lines and the PSTN line. In this case, 1 digit should be
stripped before the call is placed.
Prepend
Specify the digits to be prepended before the call is placed via the trunk. Those
digits will be prepended after the dialing number is stripped.
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Time Condition
Users could customize holiday time, office time or a specified time to allow the
Time Condition
outbound route to be used.
Outbound Blacklist
The UCM6200 allows users to configure blacklist for outbound routes. If the dialing number matches the blacklist
numbers or patterns, the outbound call will not be allowed. The outbound blacklist can be configured under UCM
Web GUIExtension/TrunkOutbound Routes: Outbound Blacklist.
Users can configure numbers, patterns or select country code to add in the blacklist. Please note that the blacklist
settings apply to all outbound routes.
Figure 110: Country Codes
PIN Groups
The UCM6200 supports pin group. Once this feature is configured, users can apply pin group to specific
outbound routes. When placing a call on pin protected outbound routes, caller will be asked to input the group
pin number, this feature can be found on the webGUIExtension/TrunkOutbound RoutesPIN Groups.
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Table 58: Outbound Routes/PIN Group
Name
Specify the name of the group
Record In CDR
Specify whether to enable/disable record in CDR
PIN Number
Specify the code that will asked once dialing via a trunk
PIN Name
Specify the name of the PIN
Once user click on
the following figure shows to configure the new PIN.
Figure 111: Create New PIN Group
The following screenshot shows an example of created PIN Groups and members:
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Figure 112: PIN Members
Note:
If PIN group is enabled on outbound route level, password, privilege level and enable filter on source caller ID
will be disabled.
Figure 113: Outbound PIN
If pin group CDR is enabled, the call with PIN group information will be displayed as part of CDR under Account
Code field.
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Figure 114: CDR Record
- Importing PIN Groups from CSV files:
User can also import PIN Groups by uploading CSV files for each group. To do this:
1. Navigate to Extension/TrunkOutbound RoutesPIN Groups and click on the “Choose file to
upload” button.
Figure 115: Importing PIN Groups from CSV files
2. Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in
error messages such as the one below:
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Figure 116: Incorrect CSV File
3. To ensure a successful import, please follow the format in the sample image below
Figure 117: CSV File Format
•
The top-left value (A1) is the PIN Group name. In this case, it is “ALPHA”.
•
Row 2 contains the labels for the modifiable fields: pin and pin_name. These values should not
be changed and will cause an upload error otherwise.
•
Rows 3+ contain the user-defined values with Column A holding the PINs and Column B
holding the PIN names. PIN values must consist of at least four digits.
•
Once the file is successfully uploaded, the entry will be added to the list of PIN Groups.
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Figure 118: CSV File Successful Upload
Inbound Routes
Inbound routes can be configured via Web GUIExtension/TrunkInbound Routes.
•
Click on
•
Click on "Blacklist" to configure blacklist for all inbound routes.
•
Click on
to edit the inbound route.
•
Click on
to delete the inbound route.
to add a new inbound route.
Inbound Rule Configurations
Table 59: Inbound Rule Configuration Parameters
Trunks
Select the trunk to configure the inbound rule.
•
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.
Pattern
Example: [12345-9] - Any digit from 1 to 9.
•
The pattern can be composed of two parts, Pattern and CallerID Pattern. The
first part is used to specify the dialed number while the second part is used to
specify the caller ID and it is optional, if set it means only the extension with
the specific caller ID can call in or call out.
For example, pattern
'_2XXX/1234' means the only extension with the caller ID '1234' can use this
rule.
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Notes:
▪
Multiple patterns can be used. Each pattern should be entered in new line.
▪
Users can add comments to a dial plan by typing “/*” and “*/” before and after
each comment respectively.
▪
Example:
Pattern
CallerID Pattern
_X.
1000
_NNXXNXXXXX /* 10-digit long distance */
1001
_818X. /* Any number with leading 818 */
After creating the inbound route, users can choose to enable and disable it. If the
Disable This Route
route is disabled, it will not take effect anymore. However, the route settings will
remain in UCM. Users can enable it again when it’s needed.
Prepend Trunk Name
Prepend trunk name to display.
Configure the Alert-Info, when UCM6200 receives an INVITE request, the Alert-Info
Alert-Info
header field specifies an alternative ring tone to the UAS.
Multiple mode allows user to switch between destinations of the inbound rule by
Inbound Multiple
feature codes. Configure related feature codes as described in [Inbound Route:
Mode
Multiple Mode]. If this option is enabled, user can use feature code to switch
between different destinations.
Select the default destination for the inbound call.
Default Destination
•
Extension
•
Voicemail
•
Conference Room
•
Queue
•
Ring Group
•
Paging/Intercom
•
Voicemail Group
•
Fax
•
DISA
•
IVR
•
Dial by Name
•
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.
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Configure the number of digits to be stripped from the beginning of the DID. This
Strip
option shows up only when "By DID" is selected.
Configure the number of digits to be prepended to an inbound DID pattern, with
Prepend
strip taking precedence over prepend.
This option shows up only when "By DID" is selected. If enabled, the external users
Dial Trunk
dialing in to the trunk via this inbound route can dial outbound call using the
UCM6200’s trunk.
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:
DID Destination
•
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 to route calls to different extensions. The following example demonstrates the process:
1. If Trunk provides a DID pattern of 18005251163.
2. If Strip is set to 8, 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.
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Figure 119: Inbound Route feature: Prepend
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 120: Inbound Route - Multiple Mode
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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 GUI login.
For example, the customer service hotline destination has to be set to a different IVR after 7PM. The user can
dial *62 to switch to “Mode 1” with that IVR set as the destination before off work.
To customize feature codes for “Default Mode” and “Mode 1”, click on
under
“Inbound Routes” page, check “Enable Inbound Multiple Mode” option and change “Inbound Default Mode” and
“Inbound Mode 1” values (By default, *61 and *62 respectively).
Figure 121: Inbound Route - Multiple Mode Feature Codes
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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.
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. Anonymous can also
be added as a Blacklist Number.
•
To remove a number from the Blacklist, select the number in "Blacklist list" and click on
.
Figure 122: Blacklist Configuration Parameters
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•
To add blacklist number in batch, click on “choose file to upload” to upload blacklist file in csv format. The
supported csv format is as below.
Figure 123: Blacklist csv File
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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Call FeaturesFeature Codes.
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CONFERENCE ROOM
The UCM6200 supports conference room allowing multiple rooms used at the same time:
•
UCM6202/6204 supports up to 3 conference rooms allowing up to 25 simultaneous PSTN or IP participants.
•
UCM6208 supports up to 6 conference rooms allowing up to 32 simultaneous PSTN or IP participants.
The conference room configurations can be accessed under Web GUICall FeaturesConference. In this
page, users could create, edit, view, invite, manage the participants and delete conference rooms. The
conference room status and conference call recordings (if recording is enabled) will be displayed in this web
page as well.
Conference Room Configurations
•
Click on "Create New Conference Room" to add a new conference room.
•
Click on
to edit the conference room.
•
Click on
to delete the conference room.
Table 60: Conference Room Configuration Parameters
Extension
Configure the conference number for the users to dial into the conference.
When configured, the users who would like to join the conference call must enter
this password before accessing the conference room.
Note:
Password
•
If "Public Mode" is enabled, the password is not required to join the conference
room thus this field is invalid.
•
The password must be at least 4 characters.
Configure the password to join the conference room 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
Admin Password
the invited party) during the conference call.
Note:
•
If "Public Mode" is enabled, the password is not required to join the conference
room thus this field is invalid.
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•
The password must be at least 4 characters.
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Enable Caller Menu
If enabled, conference participant could press the * key to access the conference
room menu. The default setting is "No".
If enabled, the calls in this conference room will be recorded automatically in a
Record Conference
.wav format file. All the recording files will be displayed and can be downloaded in
the conference web page. The default setting is "No".
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".
Quiet Mode
Note:
"Quiet Mode" and "Announce Callers" cannot be enabled at the same time.
If enabled, the participants will not hear each other until the conference
administrator joins the conference. The default setting is "No".
Wait For Admin
Note:
If "Quiet Mode" is enabled, the voice prompt for "Wait For Admin" will not be
announced.
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.
Enable User Invite
The default setting is "No".
Note:
Conference administrator can always invite other users without enabling this
option.
If enabled, the caller will be announced to all conference participants when there
Announce Callers
the caller joins the conference. The default setting is "No".
Note:
"Quiet Mode" and "Announce Callers" cannot be enabled at the same time.
If enabled, no authentication will be required when joining the conference call. The
Public Mode
default setting is "Yes".
If enabled, the UCM6200 will play Hold music when there is only one user in the
Play Hold Music
conference. The default setting is "No".
Select the music on hold class to be played in conference call. Music On Hold
Music On Hold
class can be set up under Web GUIPBX SettingsMusic On Hold.
Skip Authentication
When Inviting User
If enabled, the invitation from Web GUI for a conference room with password will
via Trunk from Web
skip the authentication for the invited users. The default setting is "No".
GUI
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Conference Settings contains the following options:
Table 61: Conference Settings
Enable Talk detection
If enabled, the AMI will send the corresponding event when a user
starts or ends talking.
DSP Talking Threshold
The time in milliseconds of sound above what the dsp has
established as base line silence for a user before a user is
considered to be talking. This value affects several operations and
should not be changed unless the impact on call quality is fully
understood, the default value is 128.
DSP Silence Threshold
The time in milliseconds of sound falling within the what the dsp
has established as base line silence before a user is considered
to be silent. This value affects several operations and should not
be changed unless the impact on call quality is fully understood,
the default value is 2500.
Users can check the talking Caller IDs in conference control page (UCM WebUICall FeaturesConference).
The image will move up and down when the user is talking.
Figure 124: Conference
Join a Conference Call
Users could dial the conference room extension to join the conference. If password is required, enter the
password to join the conference as a normal user, or enter the admin password to join the conference as
administrator.
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Invite Other Parties to Join Conference
When using the UCM6200 conference room., there are two ways to invite other parties to join the conference.
•
Invite from Web GUI.
For each conference room in UCM6200 Web GUI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 125: Conference Invitation from Web GUI
•
Invite by dialing 0 or 1 during conference call.
A conference participant can invite other parties to the conference by dialing from the phone during the
conference call. Please make sure option "Enable User Invite" is turned on for the conference room first. Enter
0 or 1 during the conference call. Follow the voice prompt to input the number of the party you would like to
invite. A call will be sent to this number to join it into the conference.
0: If 0 is entered to invite other party, once the invited party picks up the invitation call, a permission will be asked
to "accept" or "reject" the invitation before joining the conference.
1: If 1 is entered to invite other party, no permission will be required from the invited party.
--------------------------------------------------------------------------------------------------------------------------------------------------Note:
Conference administrator can always invite other parties from the phone during the call by entering 0 or 1. To
join a conference room as administrator, enter the admin password when joining the conference. A conference
room can have multiple administrators.
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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 room will be listed.
1. Click on
to kick a participant from the conference.
2. Click on
to mute the participant.
3. Click on
to lock this conference room so that other users cannot join it anymore.
4. Click on
•
to invite other users into the conference room.
Manage the conference call from IVR.
If "Enable Caller Menu" is enabled, conference participant can input * to enter the IVR menu for the conference.
Please see options listed in the table below.
Table 62: Conference Caller IVR Menu
Conference Administrator IVR Menu
1
Mute/unmute yourself.
2
Lock/unlock the conference room.
3
Kick the last joined user from the conference.
4
Decrease the volume of the conference call.
5
Decrease your volume.
6
Increase the volume of the conference call.
7
Increase your volume.
More options.
8
•
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.
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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.
----------------------------------------------------------------------------------------------------------------------------- -------------------Note:
When there is participant in the conference, the conference room configuration cannot be modified.
----------------------------------------------------------------------------------------------------------------------------- --------------------
Record Conference
The UCM6200 allows users to record the conference call and retrieve the recording from Web GUICall
FeaturesConferenceRecord Conference.
To record the conference call, when the conference room is in idle, enable "Record Conference" from the
conference room configuration dialog. Save the setting and apply the change. When the conference call starts,
the call will be automatically recorded in .wav format.
The recording files will be listed as below once available. Users could click on
click on
to delete the recording.
to download the recording or
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.
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Figure 126: Conference Recording
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CONFERENCE SCHEDULE
Conference Schedule Configuration
Conference Schedule can be found under UCM6200 Web GUICall FeaturesConference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 63: Conference Schedule Parameters
Schedule Options
Conference Topic
Conference Room
Configure the name of the scheduled conference. Letters, digits, _ and - are
allowed.
Select a conference room for this scheduled conference.
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,
Kick Time(m)
this conference room will be cleared and locked for the scheduled conference to
begin.
Note: Kick Time cannot be less than 6 minutes 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.
Configure the beginning date and duration of scheduled conference.
Schedule Time
Note: Please pay attention to avoid time conflict on schedules in the same
conference room.
Enable Google
Calendar
Conference
Administrator
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].
Select the administrator of scheduled conference from selected extensions.
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“Public Mode” must be disabled from Conference Room Options tab.
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Local Extension
Select available extensions from the list to attend scheduled conference.
Select available extensions from the remote peer PBX.
Remote Extension
Note:
“LDAP Sync” must be enabled on the UCM6200 to view remote extensions.
Add extensions that are not in the list (both local and remote list). If the user
Special Extension
wishes to add the special extension, please match the pattern on the outbound
route.
Remote Conference
Invite a remote conference.
Conference Room Options
Configure conference room password. Please note that if “Public Mode” is
Password
enabled, this option is automatically disabled.
Admin Password
Enable Caller Menu
Record Conference
Configure the password to join as conference administrator. Please note that if
“Public Mode” is enabled, this option is automatically disabled.
If this option is enabled, conference participants will be able to access conference
room menu by pressing the * key.
If this option is enabled, conference call will be recorded in .wav format. The
recorded file can be found from Conference page.
If this option is enabled, the notification tone or voice prompt for joining or leaving
the conference won’t be played.
Quiet Mode
Note:
Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same
time.
If this option is enabled, the participants in the conference won’t be able to hear
each other until conference administrator joins the conference.
Wait For Admin
Note:
If “Quiet Mode” is enabled, voice prompt for this option won’t be played.
If this option is enabled, the user can:
•
Press ‘0’ to invite others to join the conference with invited party’s
permission
Enable User Invite
•
Press ‘1’ to invite without invited party’s permission
•
Press ‘2’ to create a multi-conference room to another conference room
•
Press ‘3’ to drop all current multi-conference rooms.
Note:
Conference Administrator is always allowed to access this menu.
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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.
Announce Callers
Note:
Option “Quiet Mode” and option “Announce Caller” cannot be enabled at the same
time.
If this option is enabled, no authentication is required for entering the conference
room.
Public Mode
Note:
Please be aware of the potential security risks when turning on this option.
If this option is enabled, UCM6200 will play Hold Music while there is only one
Play Hold Music
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
If this option is enabled, conference schedules will be automatically cleaned as
Schedules Cleaner
configured.
Conference
Schedules Clean
Enter the clean time (in hours). The valid range is from 0 to 23.
Time
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 GUI to display scheduled conference in Conference Schedule Table. Please
see figure below.
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Figure 127: Conference Schedule
Once the conference room is scheduled, at the kick time, all users will be removed from conference room and
no extension can join the conference room anymore. At the scheduled conference time, UCM6200 will send
INVITE to the extensions that have been selected for conference.
----------------------------------------------------------------------------------------------------------------------------- ---------------------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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.
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Figure 128: Create New IVR
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Table 64: 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.
If enabled, all callers to the IVR can use trunk. The permission must be configured
Dial Trunk
for the users to use the trunk first. The default setting is "No".
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
Permission
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 allowed to go through.
This controls the destination that can be reached by the external caller via the
inbound route. The available destinations are:
Dial Other
Extensions
•
Extension
•
Conference
•
Call Queue
•
Ring Group
•
Paging/Intercom Groups
•
Voicemail Groups
•
Fax Extension
•
Dial by Name
•
All
IVR Black/White
If enabled only numbers inside of the White List or outside of the Black List can be
List
called from IVR.
Internal
Black/White List
Contain numbers, either of Black List or White List.
This feature can be used only when Dial Trunk is enabled, it contains external
External
Black/White List
Replace Caller ID
numbers allowed or denied calling from the IVR, the allowed format is the following:
Number1, number2, number3, ….
If enabled, the UCM will replace the caller display name with the IVR name the
caller know whether the call is incoming from a direct extension or an IVR.
When present in an INVITE request, the alert-Info header field specifies and
Alert Info
alternative ring tone to the UAS.
Select an audio file to play as the welcome prompt for the IVR. Click on "Prompt"
Welcome Prompt
to
add
additional
audio
file
under
Web
GUIPBX
SettingsVoice
PromptCustom Prompt.
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Configure the timeout between digit entries. After the user enters a digit, the user
Digit Timeout
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 3s.
After playing the prompts in the IVR, the UCM6200 will wait for the DTMF entry
Response
within the timeout (in seconds). If no DTMF entry is detected within the timeout, a
Timeout
timeout prompt will be played. The default setting is 10 seconds.
Response
Select the prompt message to be played when timeout occurs.
Timeout Prompt
Invalid Prompt
Select the prompt message to be played when an invalid extension is pressed.
Response
Configure the number of times to repeat the prompt if no DTMF input is detected.
Timeout Repeat
When the loop ends, it will go to the timeout destination if configured, or hang up.
Loops
The default setting is 3.
Configure the number of times to repeat the prompt if the DTMF input is invalid.
Invalid Repeat
When the loop ends, it will go to the invalid destination if configured, or hang up.
Loops
The default setting is 3.
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
SettingsVoice PromptLanguage Settings. The dropdown list shows all the
Language
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 SettingsVoice PromptLanguage Settings.
Key Pressing Events
Select the event for each key pressing for 0-9, *, Timeout and Invalid. The event
options are:
Key Press Event:
Press 0
Press 1
Press 2
Press 3
Press 4
Press 5
Press 6
Press 7
Press 8
Press 9
Press *
•
Extension
•
Voicemail
•
Conference Rooms
•
Voicemail Group
•
IVR
•
Ring Group
•
Queues
•
Page Group
•
Fax
•
Custom Prompt
•
Hangup
•
DISA
•
Dial by Name
•
External Number
•
Callback
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Timeout
Invalid
When exceeding the number of defined answer timeout, IVR will enter the
configured event when timeout. If not configured, then it will Hangup.
Configure the destination when the Invalid Repeat Loop is done.
Figure 129: Key Pressing Events
Black/White List in IVR
In some scenarios, the IPPBX administrator needs to restrict the extensions that can be reached from IVR.
For example, the company CEO and directors prefer only receiving calls transferred by the secretary, some
special extensions are used on IP surveillance end points which shouldn’t be reached from external calls via IVR
for privacy reason. UCM has now added blacklist and whitelist in IVR settings for users to manage this.
To use this feature, log in UCM Web GUI and navigate to Call FeaturesIVRCreate/Edit IVR: IVR
Black/White List.
•
If the user selects “Blacklist Enable” and adds extension in the list, the extensions in the list will not be
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•
If the user selects “Whitelist Enable” and adds extension in the list, only the extensions in the list can be
allowed to be reached via IVR.
Figure 130: Black/White List
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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
SettingsVoice PromptCustom Prompt page directly.
Figure 131: 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.
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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 SettingsVoice
PromptLanguage Settings.
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 132: Language Settings for Voice Prompt
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A new dialog window of voice prompt package list will be displayed. Users can see the version number (latest
version available V.S. current installed version), package size and options to upgrade or download the language.
Figure 133: 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 134: New Voice Prompt Language Added
A new language option will be displayed after successfully installed. Users then could select it to apply in the
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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 GUIPBX SettingsVoice PromptLanguage Settings instead of the entire
language pack.
Figure 135: 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Call FeaturesVoicemail.
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Figure 136: Voicemail Settings
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Table 65: Voicemail Settings
Configure the maximum number of seconds for the voicemail greeting. The
Max Greeting
default setting is 60 seconds.
Dial ‘0’ For Operator
Operator Extension
If enabled, the caller can press 0 to exit the voicemail application and connect
to the configured operator’s extension.
Select the operator extension, which will be dialed when users press 0 to exit
voicemail application. The operator extension can also be used in IVR.
Max Messages Per
Configure the maximum number of messages per folder in users’ voicemail. The
Folder
valid range 10 to 1000. The default setting is 50.
Select the maximum duration of the voicemail message. The message will not
be recorded if the duration exceeds the max message time. The default setting
is 15 minutes. The available options are:
Max Message Time
•
1 minute
•
2 minutes
•
5 minutes
•
15 minutes
•
30 minutes
•
Unlimited
Configure the minimum duration (in seconds) of a voicemail message.
Messages will be automatically deleted if the duration is shorter than the Min
Message Time. The default setting is 3 seconds. The available options are:
•
No minimum
Min Effective Message
•
1 second
Time
•
2 seconds
•
3 seconds
•
4 seconds
•
5 seconds
Note: Silence and noise duration are not counted in message time.
Announce Message
If enabled, the caller ID of the user who has left the message will be announced
Caller-ID
at the beginning of the voicemail message. The default setting is "No".
Announce Message
If enabled, the message duration will be announced at the beginning of the
Duration
voicemail message. The default setting is "No".
If enabled, a brief introduction (received time, received from, and etc) of each
Play Envelope
message will be played when accessed from the voicemail application. The
default setting is "Yes".
If enabled, UCM will play from the voice message left most recently; if disabled,
Play from Last
UCM will play from the earliest left voice message
Allow User Review
If enabled, users can review the message following the IVR before sending.
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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 prompted to enter the voicemail
password and then can enter digits from the phone keypad to navigate in the IVR menu for different options.
Table 66: 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
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
1 - Accept this recording
message
2 - Listen to it
3 - Re-record your message
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2 - Record your busy message
1 - Accept this recording
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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
Extension Voicemail Count
The UCM62xx provides an easy way to check the number of voicemail messages for each extension directly
from UCM web GUI  Extension/Trunk  Extensions overview page.
Voicemail count (“Message” column) is displayed in the format Urgent / Total / Read.
Figure 137: Voicemail Count
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.
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Table 67: Voicemail Email Settings
Attach Recordings to
If enabled, voicemails will be sent to user's Email address. The default setting
E-Mail
is "Yes".
Keep Recordings
If enabled, voicemail will be stored in the UCM6200 after the email is sent. The
default setting is “Yes”.
Fill in the "Subject:" and "Message:" content, to be used in the Email when
sending to the user.
The template variables are:
•
\t: TAB
Template for Voicemail
•
${VM_NAME}: Recipient's first name and last name
Emails
•
${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 138: 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Call
FeaturesVoicemailVoicemail Group. Click on "Create New Voicemail Group" to configure the group.
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Figure 139: Voicemail Group
Table 68: Voicemail Group Settings
Enter the Voicemail Group Extension. The voicemail messages left to this extension
Extension
will be forwarded to all the voicemail group members.
Configure the Name to identify the voicemail group. Letters, digits, _ and - are
Name
allowed.
Voicemail
Configure the voicemail password for the users to check voicemail messages.
Password
Email Address
Configure the Email address for the voicemail group extension.
Voicemail Group
Select available mailboxes from the left list and add them to the right list. The
Mailboxes
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Call FeaturesRing Group.
Figure 140: Ring Group
•
Click on
•
Click on
to edit the ring group. The following table shows the ring group configuration parameters.
•
Click on
to delete the ring group.
to add ring group.
Table 69: Ring Group Parameters
Ring Group Name
Extension
Configure ring group name to identify the ring group. Letters, digits, _ and –
are allowed.
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.
Select available remote users from the left side to the ring group member list
Selected LDAP Numbers
on the right side. Click on
be enabled first.
to arrange the order. Note: LDAP Sync must
Select the ring strategy. The default setting is “Ring in order”.
•
Ring Strategy
Ring simultaneously.
Ring all the members at the same time when there is incoming call to the
ring group extension. If any of the member answers the call, it will stop
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ringing.
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•
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.
Select the “Music On Hold” Class of this Ring Group, “Music On Hold” can be
Music On Hold
managed from the “Music On Hold” panel on the left.
This option is to set a custom prompt for a ring group to announce to caller.
Click on ‘Prompt’, it will direct to the page PBX SettingsVoice
Custom Prompt
PromptCustom Prompt, where users could record new prompt or upload
prompt files.
Configure the number of seconds to ring each member. If set to 0, it will keep
Ring Timeout on Each
Member
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.
If enabled, calls on this ring group will be automatically recorded. The default
Auto Record
setting
is
No.
The
recording
files
can
be
accessed
from
Web
GUICDRRecording Files.
If enabled, the UCM will replace the caller display name with the Ring Group
Replace Caller ID
name the caller know whether the call is incoming from a direct extension or a
Ring Group.
If enabled, users could select extension, voicemail, ring group, IVR, call queue,
Enable Destination
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.
Configure the password to access the ring group extension's voicemail.
Secret
Note:
The password must be at least 4 characters.
Configure the Email address of the ring group extension's voicemail. If "Attach
Email Address
Recordings
to
E-mail"
is
enabled
from
Web
GUIPBX
SettingsVoicemailVoicemail Email Settings, the voicemail can be sent
to the ring group's Email address as attachment.
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Figure 141: 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
GUIExtension/Trunk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
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UCM6200_A update remote LDAP server automatically from peer UCM6200_B. In addition, Sync LDAP
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Password must match for UCM6200_A and UCM6200_B 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.
Figure 142: 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.
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Figure 143: Manually Sync LDAP Server
4. Under Ring Groups setting page, click “Add”. Ring Groups can be found under Web GUI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.
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Figure 144: 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 section describes the configuration of
paging/intercom group under Web GUICall FeaturesPaging/Intercom.
Configure Paging/Intercom Group
•
Click on "Create New Paging/Intercom Group" to add paging/intercom group.
Figure 145: Paging/Intercom Group
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Table 70: 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".
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 SettingsVoice
Custom Prompt
PromptCustom Prompt, where users could record new prompt or upload
prompt files.
Page/Intercom Group
Select available users from the left side to the paging/intercom group member
Members
list on the right.
•
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 146: 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Call Features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Call FeaturesCall Queue.
Configure Call Queue
Call queue settings can be accessed via Web GUICall FeaturesCall Queue.
Figure 147: Call Queue
UCM6200 supports custom prompt feature in call queue. This custom prompt will active after the caller waits for
a period of time in the Queue. Then caller could choose to leave a message/ transfer to default extension or
keep waiting in the queue.
To configure this feature, please go to UCM Web GUICall FeaturesCall QueueCreate New Queue/Edit
QueueQueue Optionsset Enable Destination to Enter Destination with Voice Prompt. Users could configure
the wait time with Voice Prompt Cycle.
•
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.
•
Click on
to delete the call queue.
Table 71: Call Queue Configuration Parameters
Basic Settings
Extension
Configure the call queue extension number.
Name
Configure the call queue name to identify the call queue.
Select the strategy for the call queue.
Strategy
•
Ring All
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Ring all available Agents simultaneously until one answers.
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•
Linear
Ring agents in the specified order.
•
Least Recent
Ring the agent who has been called the least recently.
•
Fewest Calls
Ring the agent with the fewest completed calls.
•
Random
Ring a random agent.
•
Round Robin
Ring the agents in Round Robin scheduling with memory.
The default setting is "Ring All".
Select the Music On Hold class for the call queue.
Music On Hold
Note: Music On Hold classes can be managed from Web GUIPBX
SettingsMusic On Hold.
Configure the maximum number of calls to be queued at once. This number does
not include calls that have been connected with agents. It only includes calls not
Max Queue Length
connected yet. The default setting is 0, which means unlimited. When the
maximum value is reached, the caller will be treated with busy tone followed by
the next calling rule after attempting to enter the queue.
Configure the number of seconds before a new call can ring the queue after the
Wrapup Time
last call on the agent is completed. If set to 0, there will be no delay between calls
to the queue. The default setting is 10 seconds.
Retry Time
Configure the number of seconds to wait before ringing the next agent.
Configure the number of seconds an agent will ring before the call goes to the
Ring Time
next agent. The default setting is 30 seconds.
If enabled, the calls on the call queue will be automatically recorded. The
Auto Record
recording files can be accessed in Queue Recordings under Web GUICall
FeaturesCall Queue.
Configure the timeout after which users will be disconnected from the call queue.
Max Wait Time
The default setting is "0" which means unlimited.
Note: It is recommended to configure "Wait Time" longer than the "Wrapup Time".
Once Max Wait Time has been configured, select to which destination send the
Destination
calls that have timed out. The default is to “Hang up” the call.
Destination Prompt
Cycle
Configure the voice prompt cycle (in seconds) of the call queue. Once all agents
are busy and the voice prompt will be played and you can press the appropriate
key to transfer to failover destination.
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Custom Prompt
When playing a custom prompt, press 1 to transfer to failover destination.
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Select failover destination to send callers after pressing 1 upon hearing the
Destination
custom prompt.
Advanced Settings
-
Virtual Queue
-
Position
Refer to Call Center Settings and enhancements section for detailed
Announcement
information about these features.
-
Queue Chairman
Enable Agent Login
Enables agent login/logout feature for static agents (supported only on GXP21XX
phones with fw higher than 1.0.9.18).
Configure whether the callers will be disconnected from the queue or not if the
queue has no agent anymore. The default setting is "Strict".
•
Yes
Callers will be disconnected from the queue if all agents are paused or invalid.
Leave When Empty
•
No
Never disconnect the callers from the queue when the queue is empty.
•
Strict
Callers will be disconnected from the queue if all agents are paused, invalid
or unavailable.
Configure whether the callers can dial into a call queue if the queue has no agent.
The default setting is "No".
•
Yes
Callers can always dial into a call queue.
Dial in Empty 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.
Report Hold Time
Replace Display Name
Enable Feature Codes
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".
If enabled, the UCM will replace the caller display name with the Call Queue name
so that the caller knows the call is incoming from a Call Queue.
Enable feature codes option for call queue. For example, *83 is used for “Agent
Pause”
Dynamic Login
If enabled, the configured PIN number is required for dynamic agent to log in. The
Password
default setting is disabled.
Alert-Info
Configure the call destination for the call to be routed to if no agent in this call
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queue answers the call.
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Agents
Go to “Agents” Tab and Select the available users to be the static agents in the
call queue. Choose from the available users on the left to the static agents list on
Agents
the right. Click on
to choose. And use UP and Down arrow to select the
order of the agent within the call queue.
Static Agents limitation:
To guarantee a high level of audio quality with the call queue feature, UCMs will limit the number of static agents
allowed to be assigned depending on the UCM model used. If the user attempts to configure the number of static
agents to be more than the maximum allowed number, a warning message will appear.
Figure 148: Static Agents limitation
The following table lists the maximum number of static agents for each UCM model:
Table 72: Static Agent Limitation
UCM Model
Max Static Agents in Call Queue
UCM6102
22
UCM6202
23
UCM6104
33
UCM6204
34
UCM6108 and UCM6116
45
UCM6208
75
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UCM6510
150
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Click on "Dynamic 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.
Figure 149: 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Call
Features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 under “Queue Recordings” Tab. Click on
download the recording file in .wav format; click on
to
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.
Call Center Settings and enhancements
UCM supports light weight call center features including virtual queue and position announcement, allowing the
callers to know their position on the call queue and giving them the option to either stay on the line waiting for
their turn or activate a callback which will be initiated by the UCM one an agent is free.
To configure call center features, press
on an existing call queue and go under the advanced settings tab.
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Following parameters are available:
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Table 73: Call Center Parameters
Enable Virtual Queue
Virtual Queue Period
Enable virtual queue to activate call center features.
Configure the time in (s) after which the virtual queue will take effect and the menu
will be presented to the caller to choose an option. Default is 20s.
Offered to caller after timeout: After the virtual queue period passes, the caller
will enter the virtual call queue and be presented with a menu to choose an option,
the choices are summarized below:
Virtual Queue Mode
•
Press * to set current number as callback number.
•
Press 0 to set a callback number different than current caller number.
•
Press # to keep waiting on the call queue.
Triggered on user request: In this mode, the callers can activate the virtual
queue by pressing 2, then they will be presented with the menu to choose an
option as below:
Virtual Queue
Outbound Prefix
Enable Position
Announcement
Position
Announcement
Interval
CTI Chairman
•
Press * to set current number as callback number.
•
Press 0 to set a callback number different than current caller number.
•
Press # to keep waiting on the call queue.
System will add this prefix to dialed numbers when calling back users.
Enable the announcement of the caller’s position periodically.
Configure the period of time in (s) during which the UCM will announce the caller’s
position in the call queue.
Select the extension to act as chairman of the queue (monitoring).
When enabled, statics agents can conveniently log in and out of a queue by
configuring a programmable key on their phones as a shortcut.
Notes:
✓
This feature is currently available only for GXP21xx phones on firmware
1.0.9.18 or greater.
Enable Agent Login
✓
After enabling the feature, users need to set the option on GXP21XX phone
under “AccountSIP SettingsAdvanced FeaturesSpecial Feature”
to “UCM Call Center”.
A softkey labeled “UCM-CC” will appear on the
bottom of the phone’s screen.
✓
When this option is enabled, dynamic agent login will be no longer
supported.
Queue Auto fill enhancement:
In previous UCM firmware, the call queue has a serial type behavior in that the queue will make all waiting callers
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wait in the queue even if there is more than one available member ready to take calls until the head caller is
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connected with the member they were trying to get to.
The next waiting caller in line then becomes the head caller, and they are then connected with the next available
member and all available members and waiting callers waits while this happens.
Starting from 1.0.14.x, the waiting callers are connecting with available members in a parallel fashion until there
are no more available members or no more waiting callers.
For example, in a call queue with linear method, if there are two available agents, when two callers call in the
queue at the same time, UCM will assign the two callers to each of the two available agents at the same time,
rather than assigning the second caller to second available agent after the first agent answers the call from the
first caller.
Queue Statistics
Along with call center features, users can also gather detailed call queue statistics allowing them to make better
changes/decision to manage better the call distribution and handling based on time, agent and queue.
To access call queue statistics, go to Web GUICall FeaturesCall Queue and click on “Call Queue Statistics”,
the following page will be displayed:
Figure 150: Call Queue Statistics
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Select the time interval along with the queue(s) and agent(s) to get detailed statistics, then press on which
information to be displayed (call distribution by agent, virtual queue distribution by hour...).
Switchboard
Switchboard is a Web GUI tool for call queue monitoring and management, admin can access to it from the
menu Call FeaturesCall Queue then press « Switchboard ».
Following page will be displayed:
Figure 151: Call Queue Switchboard
The table below gives a brief description for the main menus:
Table 74: Switchboard Parameters
Waiting
This menu shows the current waiting calls along with the caller id and the option
to hang-up call by pressing on the
button.
Shows the current established calls along with the caller id and the callee (agent)
Proceeding
as well as the option to hang-up, transfer, add conference or barge-in the call
using the call options buttons.
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Displays the list of agents in the queue and the extension status (idle, ringing, in
Agents
use or unavailable) along with some basic call statistics and agent’s mode (static
or dynamic).
There are three different privilege levels for Call Queue management from the switchboard: Super Admin, Queue
Chairman, and Queue Agent.
•
Super Admin - Default admin of the UCM. Call queue privileges include being able to view and edit all
queue agents, monitor and execute actions for incoming and ongoing calls for each extension in
Switchboard, and generate Call Queue reports to track performance.
•
Queue Chairman - User appointed by Super Admin to monitor and manage an assigned queue extension
via Switchboard. The Queue Chairman can log into the UCM user portal with his extension number and
assigned user password. To access the Switchboard, click on “Value-added Features” in the side menu and
click on “Call Queue”. In the image below, User 1012 is the Queue Chairman appointed to manage Queue
Extension 6500 and can see all the agents of the queue in the Switchboard.
•
Queue Agent - User appointed by Super Admin to be a member of a queue extension. A queue agent can
log into the UCM user portal with his extension number and assigned user password. To access the
Switchboard, click on “Value-added Features” in the side menu and click on “Call Queue”. However, a queue
agent can view and manage only his own calls and statistics, but not other agents’ in the queue extension.
In the image below, User 1007 is a queue agent and can see only his own information in the Switchboard.
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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Call FeaturesPickup Groups.
•
Click on
•
Click on
•
Click on
to create a new pickup group.
to edit the pickup group.
to delete the pickup group.
Select extensions from the list on the left side to the right side.
Figure 152: Edit Pickup Group
Configure Pickup Feature Code
When picking up the call for the pickup group member, the user only needs to dial the pickup feature code. It’s
not necessary to add the extension number after the pickup feature code. The pickup feature code is configurable
under Web GUICall FeaturesFeature Codes.
The default feature code for call pickup extension is *8, otherwise if the person intending to pick up the call knows
the ringing extension they can use ** followed by the extension number in order to perform the call pickup
operation.
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The following figure shows where you can customize these features codes
Figure 153: Edit Pickup Feature Code
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MUSIC ON HOLD
Music On Hold settings can be accessed via Web GUIPBX Settings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 154: Music On Hold Default Class
•
Click on "Create New MOH Class" to add a new Music On Hold class.
•
Click on
•
Click on
•
Click on
to configure the MOH class sort method to be "Alpha" or "Random" for the sound files.
next to the selected Music On Hold class to delete this Music On Hold class.
to start uploading. Users can upload:
➢
Single files with 8KHz Mono Music file, or
➢
Music on hold files in a compressed package with .tar, .tar.gz and .tgz as the suffix. The file name can
only be letters, digits or special characters -_
➢
the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire
MoH.
•
Users could also download all the music on hold files from UCM. In the Music On Hold page, click on
and the file will be downloaded to your local PC.
•
Click on
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next to the sound file to delete it from the selected Music On Hold Class.
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•
Select the sound files and click on
to delete all selected music on hold files.
Note: the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire
MoH.
The UCM6200 allows Users to select the Music on Hold file from WebGUI to play it. The UCM6200 will initiate
a call to the selected extension and play this Music on Hold file once the call is answered.
Steps to play the music on hold file:
1. Click on the
button for the Music on Hold file.
2. In the prompted window, select the extension to playback and click
.
Figure 155: Play Custom Prompt
3. The selected extension will ring.
4.
Answer the call to listen to the music playback.
Users could also record their own Music on hold to override an existing custom prompt, this can be done by
following those steps:
1. Click on
.
2. A prompt of confirmation will pop up, as shown below.
Figure 156: Information Prompt
3. Click
.
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4. In the prompted window, select the extension to playback and click
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Figure 157: Record Custom Prompt
5. Answer the call and start to record your new music on hold.
6. Hangup the call and refresh Music On Hold page then you can listen to the new recorded file.
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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 SERVER
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Call
Features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.
•
Click on
to edit the Fax extension.
•
Click on
to delete the Fax extension.
Figure 158: Fax Settings
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Table 75: FAX/T.38 Settings
Enable Error
Configure to enable Error Correction Mode (ECM) for the Fax.
Correction Mode
The default setting is "Yes".
Maximum Transfer
Rate
Configure the maximum transfer rate during the Fax rate negotiation.
The possible values are 2400, 4800, 7200, 9600, 12000 and 14400.
The default setting is 14400.
Configure the minimum transfer rate during the Fax rate negotiation. The
Minimum Transfer Rate
possible values are 2400, 4800, 7200, 9600, 12000 and 14000. The default
setting is 2400.
Configure the concurrent fax that can be sent by UCM6200. Two modes “Only”
and “More” are supported.
•
Under this mode, the UCM6200 allows only single user to send fax at a time.
Max Concurrent
Sending Fax
Only
•
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
Fax Header Information
Configure the maximum length of Fax Queue from 6 to 10.
The default setting is 6.
Adds fax header into the fax file.
Configure the Email address to send the received Fax to if user's Email address
cannot be found.
Default Email Address
Note:
The extension's Email address or the Fax's default Email address needs to be
configured in order to receive Fax from Email. If neither of them is configured,
Fax will not be received from Email.
Fill in the "Subject:" and "Message:" content, to be used in the Email when
sending the Fax to the users.
The template variables are:
Template Variables
•
${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
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Receiving Fax
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Extension/Trunk page.
4. Create or edit the analog trunk for Fax as below.
Fax Mode: Make sure "Fax Mode" option is set to "None".
Figure 159: Configure Analog Trunk without Fax Detection
5. Go to UCM6200 Web GUIExtension/Trunk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.
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Figure 160: Configure Extension for Fax Machine: FXS Extension
Figure 161: Configure Extension for Fax Machine: Analog Settings
7. Go to Web GUIExtension/TrunkInbound Routes page.
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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 162: Configure Inbound Rule for Fax
Now the Fax configuration is done. When there is an incoming Fax calling to the PSTN number for the FXO port,
it will send the Fax to the Fax machine.
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Call FeaturesFax/T.38 page. Create a new Fax extension.
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Figure 163: Create Fax Extension
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3. Go to UCM6200 Web GUIExtension/TrunkAnalog Trunks page. Create a new analog trunk. Please
make sure "Fax Detection" is set to "No".
4. Go to UCM6200 Web GUIExtension/TrunkInbound Routes page. Create a new inbound route and
set the default destination to the Fax extension.
Figure 164: 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 fax@domain.local as attachment.
Figure 165: List of Fax Files
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FAX Sending
Besides the support of Fax machines, The UCM6200 supports also sending Fax via Web GUI access. This
feature can be found on Web GUIValue-added FeaturesFax Sending page. To send fax, pre-setup for
analog trunk and outbound route is required. Please refer to [ANALOG TRUNKS], [VOIP TRUNKS] and [Outbound
Routes] sections for configuring analog trunk and outbound route.
After making sure analog trunk or VoIP Trunk is setup properly and 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 GUI. Users can also view the sending history is in
the same web page.
Figure 166: Fax Sending in Web GUI
After that you can see the ongoing sending operation on the progress bar.
Figure 167: Fax Send Progress
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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/ucm6xxx_busy_camp_on_guide.pdf
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PRESENCE
UCM does support SIP presence feature which allows users to advertise their current availability status and
willingness to receive calls, this way other users can use their phones in order to monitor the presence status of
each user and decide whether to call them or not based on their advertised availability.
This feature is different than BLF which is mainly used to monitor the dialog status for each extension (Ringing,
Idle or Busy). Instead the SIP presence module gives more options for users to choose which state they want to
put themselves in.
In order to configure the presence status of an extension from the web GUI, users can access the menu of
configuration using one of the two following methods:
•
From admin account, go under the menu Extension/TrunkExtensions and choose the desired extension
to edit then navigate to the “Features” tab.
OR
•
From the User Portal, go under the menu Basic InformationExtensions and navigate to the Features tab
to have the following options.
Figure 168: SIP Presence Configuration
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Select which status to set from the presence status selection drop list, six options are available and below is a
brief description of these states:
Table 76: SIP Presence Status
Available
The contact is online and can participate in conversations/phone calls.
Away
The contact is currently away (ex: for lunch break).
The contact has limited conversation flexibility and can only be reached
Chat
via chat.
Do Not Disturb
The Contact is on DND (Do Not Disturb) mode.
Custom Presence Status
Please enter the presence status for this mode on the Web GUI.
Unavailable
The contact is unreachable for the moment, please try to contact later.
Another option to set the presence status and which is more practical is using the feature code from the user’s
phone, one the user dials the feature code (default is *48), a prompt will be played to select which status they
want to put themselves in, by pressing the corresponding key.
The feature code can be enabled and customized from the Web GUICall FeaturesFeature Codes.
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Figure 169: SIP Presence Feature Code
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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 extension settings
page Web GUIExtension/TrunkExtensions.
To configure follow me:
1. Choose the extension and click on
.
2. Go to the Follow me tab to add destination numbers and enable the feature.
Figure 170: Edit Follow Me
3. Click on
to add local extensions or external numbers to be called after ringing the extension
selected in the first step.
4. Once created, it will be displayed on the follow me list. And you can click on
to delete the Follow Me.
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The following table shows the Follow Me configuration parameters:
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Table 77: Follow Me Settings
Enable
Configure to enable or disable Follow Me for this user.
If external number is added in the Follow Me, please make sure this option is
Skip Trunk Auth
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 users
By default, it is enabled and user will be asked to press 1 to accept the call or to
Confirm When
press 2 to reject the call after answering a Follow Me call.
Answering
If it is disabled, the Follow Me call will be established once after the user
answers.
Enable Destination
When enabled, the call will be routed to the default destination if no one in the
Follow Me extensions answers the call.
Configure the destination if no one in the Follow Me extensions answers the call.
The available options are:
Default Destination
Follow Me Numbers
•
Extension
•
Voicemail
•
Queues
•
Ring Group
•
Voicemail Group
•
IVR
•
External Number
The added numbers are listed here. Click on
on
to delete the number. Click on
to arrange the order. Click
to add new numbers.
Add a new Follow Me number which could be a ‘Local Extension’ or ‘External
New Follow Me
Number’. The selected dial plan should have permissions to dial the defined
Number
external number.
Select the order in which the Follow Me destinations will be dialed to reach the
Dialing Order
user: ring all at once or ring one after the other.
Click on “Follow Me Options” under Web GUIExtension/TrunkExtension page to enable or disable the
options listed in the following table.
Table 78: Follow Me Options
Playback Incoming
If enabled, the PBX will playback the incoming status message before starting
Status Message
the Follow Me steps.
Record the Caller’s
If enabled, the PBX will record the caller’s name from the phone so it can be
Name
announced to the callee in each step.
Playback Unreachable
If enabled, the PBX will playback the unreachable status message to the caller
Status Message
if the callee cannot be reached.
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SPEED DIAL
The UCM6200 supports Speed Dial feature that allows users to call a certain destination by pressing one or four
digits on the keypad allowing 6561 combinations to be configured. This creates a system-wide speed dial access
for all the extensions on the UCM6200.
To enable Speed Dial, on the UCM6200 Web GUI, go to page Web GUICall FeaturesSpeed Dial.
User should first click on
.
Then decide from one digit up to four digits combination used for Speed
Dial and select a dial destination from “Default Destination”. The supported destinations include extension,
voicemail, conference room, voicemail group, IVR, ring group, call queue, page group, fax, DISA, Dial by Name
and external number.
Figure 171: Speed 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
•
Click on
to edit the DISA configuration.
•
Click on
to delete the DISA.
to add a new DISA.
Figure 172: Create New DISA
The following table details the parameters to set and configure DISA feature on UCM6200 PBX.
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Table 79: DISA Settings
Name
Configure DISA name to identify the DISA.
Configure the password (digit only) required for the user to enter before using DISA
to dial out.
Password
Note:
The password must be at least 4 digits.
Configure the permission level for DISA.
The available permissions are "Internal", "Local", "National" and "International"
from the lowest level to the highest level.
The default setting is "Internal".
Permission
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.
Configure the maximum amount of time the UCM6200 will wait before hanging up
Response Timeout
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.
If enabled, during an active call, users can enter the UCM6200 Hangup feature
Allow Hangup
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".
Replace Display
Name
If enabled, the UCM will replace the caller display name with the DISA name.
Once successfully created, users can configure the inbound route destination as "DISA" or IVR key event as
"DISA". When dialing into DISA, users will be prompted with password first. After entering the correct password,
a second dial tone will be heard for the users to dial out.
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CALLBACK
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Call FeaturesCallback page and click on
. Configuration parameters are listed in the following table.
Table 80: Callback Configuration Parameters
Name
Configure a name to identify the Callback.
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
CallerID Pattern
caller can get callback after hanging up the call.
Note:
If leaving as blank, all numbers are allowed to use this callback.
Configure the prepend digits to be added at before dialing the outside number.
Outbound Prepend
The number with prepended digits will be used to match the outbound route. ‘-’
is the connection character which will be ignored.
Delay Before Callback
Configure the number of seconds to be delayed before calling back the user.
Configure the destination which the callback will direct the caller to. Two
destinations are available:
Destination
•
IVR
•
DISA
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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.
----------------------------------------------------------------------------------------------------------------------------- ---------------------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.
•
Sort selected extensions manually in the Eventlist
•
Click on
to edit the event list configuration.
•
Click on
to delete the event list.
Table 81: Event List Settings
Configure the name of this event list (for example, office_event_list). Please note
URI
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.
If LDAP sync is enabled between the UCM6200 and the peer UCM6200, the
Remote Extensions
remote extensions will be listed under "Available Extensions". If not, manually
enter the remote extensions under "Special Extensions" field.
Special Extensions
Manually enter the remote extensions in the peer/register trunk to be monitored
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in the event list. Valid format: 5000,5001,9000
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Figure 173: 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 SUBSCRIBE to the remote UCM6200 to obtain the remote extension status. When the SIP end
points register and subscribe 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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•
To configure LDAP sync, please go to UCM6200 Web GUIExtension/Trunk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Call FeaturesDial By Name.
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Figure 174: Create Dial by Name Group
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Figure 175: Configure Extension First Name and Last Name
1. Name
Enter a Name to identify the Dial by Name group.
2. Extension
Configure the direct dial extension for the Dial By Name group.
3. 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Extension/Trunk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 does not want to receive them directly.
4. Prompt Wait Time
Configure “Prompt Wait Time” for Dial By Name feature. During Dial By Name call, the caller will need to
input the first letters of First/Last name before this wait time is reached. Otherwise, timeout will occur and
the call might hang up. The timeout range is between 3 and 60 seconds.
5. 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.
6. 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.
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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.
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 176: Dial By Name Group In IVR Key Pressing Events
Figure 177: Dial By Name Group In Inbound Rule
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User Name Prompt Customization
Starting from fw 1.0.15.x, the Dial By Name feature can use the recorded name prompt of a user to announce
his/her name assigned to the dialed extension. If no name prompt greeting exists, the name will be spelt out like
in previous versions.
There are two ways to customize/set new user name prompt for an extension:
Upload User Name Prompt File from Web GUI
1. First, Users should have a pre-recorded file respecting the following format:
•
PCM encoded / 16 bits / 8000Hz mono.
•
In “.GSM” or “.WAV” format.
•
File size under 5M.
•
Filename must be set as the extension number. For example, the recorded file name 1000.wav will
be used for extension 1000.
2. Go under web GUI PBX Settings  Voice Prompt  User Name Prompt and click on
button.
3. Select the recorded file to upload it and press Save and Apply Settings.
•
Click on
•
Click on to play recorded user name prompt.
•
Select user name prompts and press
to record again the user name prompt.
to delete specific file or select multiple files for deletion using
the button
.
Record User Name via Voicemail Menu
The second option to record user name is using voicemail menu, please follow below steps:
•
Dial *98 to access the voicemail
•
After entering the desired extension and voicemail password, dial “0” to enter the recordings menu and
then “3” to record a name.
Another option is that each user can record their own name by following below steps:
•
The user dials *97 to access his/her voicemail
•
After entering the voicemail password, the user can press “0” to enter the recordings menu and then “3”
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to record his name.
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ACTIVE CALLS AND MONITOR
The active calls on the UCM6200 are displayed in Web GUISystem 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System StatusActive Calls. The following figure
shows extension 1002 is calling 1004. 1004 is ringing.
Figure 178: StatusPBX StatusActive Calls - Ringing
The following figure shows the call between 1002 and 1003 is established.
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Figure 179: StatusPBX StatusActive Calls – Call Established
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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 180: 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 82: 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 82: UCM6200 Feature Codes
Feature Maps
•
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.
•
Blind 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.
•
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
Attended Transfer
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.
•
Default code: *44 (Disabled by default).
•
Seamless Transfer allows user to perform blind transfer using
UCM feature code without having music on hold presented
during the transfer process, it minimizes the interruption during
Seamless Transfer
transfer, making the process smooth and simple.
•
During an active call use the feature code (*44 by default)
followed by the number you want to transfer to in order to
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•
Default code: *0
•
Enter the code during active call. It will disconnect the call.
•
Options:
Disconnect
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.
•
Default code: #72
•
Enter the code during active call to park the call.
•
Options:
Call Park
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.
•
Default code: *3
•
Enter the code followed by # or SEND to start recording the
audio call and the UCM6200 will mix the streams natively on the
fly as the call is in progress.
Audio Mix Record
•
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
•
Default Code: *90
•
Enter the code and follow the voice prompt. Or enter the code
Call Forward Busy Activate
followed by the extension to forward the call.
Call Forward Busy Deactivate
Call Forward No Answer Activate
•
Default Code: *91
•
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
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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
•
Default Setting: 1000
•
Configure the maximum interval (in milliseconds) between the
Feature Misc
Feature Code Digits Timeout
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.
•
If checked, the parking lot number can be used as extension.
The user can transfer the call to the parking lot number to park
Use Parklot as Extension
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.
Music On Hold Classes
Select the Music on Hold Class.
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
Agent Pause
Agent Unpause
•
Default Code: *97
•
Press *97 to access the voicemail box.
•
Default Code: *83
•
Pause the agent in all call queues.
•
Default Code: *84
•
Unpause the agent in all call queues.
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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.
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 In-call
•
Default Code: *45 (Disabled by default).
•
If “Pickup In-call” feature is enabled, only the extensions added
in “Allowed to seamless transfer” in the extension’s Seamless
Transfer Privilege Control List” can pick up the call.
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.
•
Default Code: *
•
This code is for the user to directly dial or transfer to an
extension's voicemail.
Direct Dial Voicemail Prefix
•
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.
Direct Dial Mobile Phone Prefix
•
Default Code: *88
•
If you have the permission to call mobile phone number, use
this prefix plus the extension number can dial the mobile phone
number of this extension directly.
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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. Disabled by default.
Listen Spy
•
Default Code: *54 (“Enable Spy” needs to be checked)
•
This is the feature code to listen in on a call to monitor
performance. Monitor’s line will be muted, and neither party will
hear from the monitor’s extension.
Whisper Spy
•
Default Code: *55 (“Enable Spy” needs to be checked)
•
This is the feature code to speak to one side of the call (for
example, whisper to employees to help them handle a call).
Only one side will be able to hear from the monitor’s extension.
Barge Spy
•
Default Code: *56 (“Enable Spy” needs to be checked)
•
This is the feature code to join in on the call to assist both
parties.
Wakeup Service
•
Default Code: *36
•
Dial this code to access UCM wakeup service, you can add,
update, activate or deactivate wakeup service.
PMS Wakeup Service
•
Default Code: *35
•
Dial this code to access UCM PMS wakeup service, you can
add, update, activate or deactivate PMS wakeup service.
Update PMS Room Status
•
Default Code: *23
•
Use this code with maid code to update PMS room status.
Choose the status to set after hearing the prompt, for example:
for maid 001 dial *23001 and then 1 after hearing the prompt.
•
Dial this code to set the presence status of the extension.
•
Possible options are:
1:"unavailable"
Presence Status
2:"available"
3:"away"
4:"chat
5:"dnd
6:"userdef"
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The UCM6200 also allows user to one click enable / disable specific feature code as shown below:
Figure 181: Enable/Disable Feature codes
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CDR. Click on
click on
to show and play the recording or
to download the recording file.
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Figure 182: Download Recording File from CDR Page
The above recorded call's recording files are also listed under the UCM6200 Web GUICDRRecording Files.
Figure 183: 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 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.
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Enable Spy
If “Enable Spy” option is enabled, feature codes for Listen Spy, Whisper Spy and Barge Spy are available for
users to dial from any extension to perform the corresponding actions.
Assume a call is on-going between extension A and extension B, user could dial the feature code from extension
C to listen on their call (*54 by default), whisper to one side (*55 by default), or barge into the call (*56 by default).
Then the user will be asked to enter the number to call, which should be either side of the active call, extension
A or B in this example.
--------------------------------------------------------------------------------------------------------------------------------------------------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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PBX SETTINGS
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 SettingsGeneral Settings.
General Settings
Table 83: Internal Options/General
General Preferences
Configure the global CallerID used for all outbound calls when no other CallerID
Global Outbound CID
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.
Configure the number of seconds to ring an extension before the call goes to
the user's voicemail box. The default setting is 60.
Ring Timeout
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.
If enabled, users will hear voice prompt before recording is started or stopped.
Record Prompt
For example, before recording, the UCM6200 will play voice prompt "The call
will be recorded". The default setting is "No".
Extension Preferences
If enabled, strong password will be enforced for the password created on the
UCM6200. The default setting is enabled.
Enforce Strong
Passwords
Strong Password Rules:
1. Password for voicemail, voicemail group, outbound route, DISA, call queue
and conference requires non-repetitive and non-sequential digits, with a
minimum length of 4 digits. Repetitive digits pattern (such as 0000, 1111,
1234, 2345, and etc), or common digits pattern (such as 111222, 321321
and etc) are not allowed to be configured as password.
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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
If enabled, random password will be generated when the extension is created.
Password
The default setting is "Yes". It is recommended to enable it for security purpose.
Enable Auto E-mail
If enabled, UCM6200 will send Email notification to user automatically after
Notification
editing extension settings or adding a new extension.
If set to "Yes", users could disable the extension range preconfigured/configured on the UCM6200. The default setting is "No".
Disable Extension
Range
Note:
It is recommended to keep the system assignment to avoid inappropriate usage
and unnecessary issues.
The default extension range assignment is:
•
User Extensions: 1000-6299
User Extensions is referring to the extensions created under Web
GUIExtension/Trunk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
Extension Ranges
settings is selected, the extension list defined in "Pick Extension Segment"
will be sent out to the device after receiving the device's request. This "Pick
Extension Segment" should be a subset of the "Pick Extensions" range
here. This feature is for the GXP series phones that support selecting
extension to be provisioned via phone's LCD.
•
Auto Provision Extensions: 5000-6299
This sets the range for "Zero Config Extension Segment" which is the
extensions can be assigned on the UCM6200 to provision the end device.
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•
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
Voice Prompt Customization
Record New Custom Prompt
Settings
In the UCM6200 Web GUIPBX SettingsVoice PromptCustom Prompt page, click on “Record New
Custom Prompt” and follow the steps below to record new IVR prompt.
Figure 184: Record New Custom Prompt
1. Specify the IVR file name.
2. Select the format (GSM or WAV) for the IVR prompt file to be recorded.
3. Select the extension to receive the call from the UCM6200 to record the IVR prompt.
4. 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.
5. Pick up the call from the extension and start the recording following the voice prompt.
6. The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play or delete
the recording.
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Upload Custom Prompt
If the user has a pre-recorded IVR prompt file, click on “Upload Custom Prompt” in Web GUIPBX
SettingsVoice Prompt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 185: Upload Custom Prompt
Click on “choose file to upload” to start uploading. Once uploaded, the file will appear in the Custom Prompt web
page.
Download All Custom Prompt
On the UCM62XX, the users can download all custom prompts from UCM Web GUI to local PC. To download
all custom prompt, log in UCM Web GUI and navigate to PBX SettingsVoice PromptCustom Prompt and
click on
. The following window will pop up in order to set a name for the
downloaded file.
Figure 186: Download All Custom Prompt
Note: The downloaded file will have a .tar extension.
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PBX Settings/Jitter Buffer
Table 84: 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".
Configure the time (in ms) to buffer. This is the jitter buffer size used in "Fixed"
Jitter Buffer Size
jitter buffer, or used as the initial time for "adaptive" jitter buffer. The default
setting is 100.
Configure the maximum time (in ms) to buffer for "Adaptive" jitter buffer
Max Jitter Buffer
implementation, or used as the jitter buffer size for "Fixed" jitter buffer
implementation. The default setting is 200.
Configure the jitter buffer implementation on the sending side of a SIP channel.
The default setting is "Fixed".
•
Implementation
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".
PBX Settings/RTP Settings
Table 85: 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.
Configure to enable or disable strict RTP protection. If enabled, RTP packets
Strict RTP
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".
Configure whether to support ICE. The default setting is enabled.
ICE is the integrated use of STUN and TURN structure to provide reliable VoIP
ICE Support
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.
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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.
STUN Server
Valid format:
[(hostname | IP-address) [':' port]
The default port number is 3478 if not specified.
PBX Settings/Payload
The UCM6200 payload type for audio codecs and video codes can be configured here.
Table 86: Internal Options/Payload
Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword packing).
AAL2-G.726
The default setting is 112.
DTMF
Configured payload type for DTMF. The default setting is 101.
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 SettingsIAX Settings.
IAX Settings/General
Table 87: IAX Settings/General
Configure the port number that the IAX2 will be allowed to listen to. The default
Bind Port
setting is 4569.
Configure the address that the IAX2 will be forced to bind to. The default setting
Bind Address
is 0.0.0.0, which means all addresses.
IAX1 Compatibility
Select to configure IAX1 compatibility. The default setting is "No".
If selected, UDP checksums will be disabled and no checksums will be
No Checksums
calculated/checked on systems supporting this feature. The default setting is
"No".
If enabled, the IAX2 will delay the rejection of calls to avoid DOS. The default
Delay Reject
setting is "No".
ADSI
Select to enable ADSI phone compatibility. The default setting is "No".
Specify which Music On Hold class this channel would like to listen to when being
Music On Hold
put on hold. This music class is only effective if this channel has no music class
Interpret
configured and the bridged channel putting the call on hold has no "Music On
Hold Suggest" setting.
Music On Hold
Specify which Music On Hold class to suggest to the bridged channel when
Suggest
putting the call on hold.
Bandwidth
Configure the bandwidth for IAX settings. The default setting is "Low".
IAX Settings/Registration
Table 88: IAX Settings/Registration
IAX Registration Options
Configure the minimum period (in seconds) of registration. The default setting is
Min Reg Expire
60.
Configure the maximum period (in seconds) of registration. The default setting is
Max Reg Expire
3600.
IAX Thread Count
Configure the number of IAX helper threads. The default setting is 10.
IAX Max Thread Count
Configure the maximum number of IAX threads allowed. The default is 100.
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If set to "yes", the connection will be terminated if ACK for the NEW message is
Auto Kill
not received within 2000ms. Users could also specify number (in milliseconds) in
addition to "yes" and "no". The default setting is "yes".
Authentication
If enabled, authentication traffic in debugging will not show. The default is "No".
Debugging
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.
•
Codec Priority
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 is 20.
Trunk Time Stamps
If enabled, time stamps will be attached to trunk frames. The default is "No".
IAX Settings/Security
Table 89: IAX Settings/Static Defense
Call Token Optional
Max Call Numbers
Max Unvalidated Call
Numbers
Call Number Limits
IP or IP Range
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.
Configure the maximum number of calls allowed for a single IP address.
Configure the maximum number of Unvalidated calls for all IP addresses.
Configure to limit the number of calls for a give IP address of 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 SettingsSIP Settings.
SIP Settings/General
Table 90: SIP Settings/General
Realm For Digest
Configure the host name or domain name for the UCM6200. Realms MUST be
Authentication
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.
Configure the IP address to bind to. The default setting is 0.0.0.0, which means
Bind IP Address
binding to all addresses.
If enabled, the UCM6200 allows unauthorized INVITE coming into the PBX and
the call can be made. The default setting is "No".
Allow Guest Calls
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.
If set to "No", all transfers initiated by the endpoint in the UCM6200 will be
Allow Transfer
disabled (unless enabled in peers or users). The default setting is "Yes".
When sending MWI NOTIFY requests, this value will be used in the "From:"
MWI 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.
Enable Diversion
Header
If disabled, the UCM will not forward the diversion header.
SIP Settings/MISC
Table 91: SIP Settings/Misc
Outbound SIP Registrations
Register Timeout
Configure the register retry timeout (in seconds). The default setting is 20.
Configure the number of registration attempts before the UCM6200 gives up. The
Register Attempts
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.
Support SIP Video
Select to enable video support in SIP calls. The default setting is "Yes".
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If enabled, when rejecting an incoming INVITE or REGISTER request, the
Reject Non-Matching
INVITE
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".
SDP Attribute Passthrough
Enable Attribute
If enable, and if the service doesn’t know the attribute of FEC/FECC/BFCP, then
Passthrough
the attribute will be passthrough.
Early Media
Enable Use Final SDP
If enabled, call negotiation will use final response SDP.
Blind Transfer
Allow callback when
blind transfer fails
Blind transfer timeout
If enabled, the UCM will callback to the transferrer when blind transfer fails
(reason of failure includes: busy and no answer).
Note: This feature takes effect only on internal calls.
Configure the timeout in (s) for the transferrer waiting for the destination to
answer. Default is 60s.
SIP Settings/Session Timer
Table 92: SIP Settings/Session Timer
Select the session timer mode. The default setting is "Accept".
The options are:
•
Originate
Always request and run session timer.
Session Timers
•
Accept
Run session timer only when requested by another UA.
•
Refuse
Do not run session timer.
Configure the maximum session refresh interval (in seconds).
Session Expire
The default setting is 1800.
Configure the minimum session refresh interval (in seconds).
Min SE
The default setting is 90.
Session Refresher
Select the session refresher to be UAC or UAS.
The default setting is UAC.
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SIP Settings/TCP and TLS
Table 93: SIP Settings/TCP and TLS
Configure to allow incoming TCP connections with the UCM6200. The default
TCP Enable
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.
Configure to allow incoming TLS connections with the UCM6200. The default
TLS Enable
setting is "No".
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.
Note:
TLS Bind Address
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
TLS Do Not Verify
Select the TLS protocol for outbound client connections. The default setting is
TLSv1.
If enabled, the TLS server's certificate won't be verified when acting as a client.
The default setting is "Yes".
This is the CA certificate if the TLS server being connected to requires self-signed
certificate, including server's public key. This file will be renamed as "TLS.ca"
TLS Self-Signed CA
automatically.
Note:
The size of the uploaded ca file must be under 2MB.
This is the Certificate file (*.pem format only) used for TLS connections. It contains
private key for client and signed certificate for the server. This file will be renamed
TLS Cert
as "TLS.pem" automatically.
Note:
The size of the uploaded certificate file must be under 2MB.
This file must be named with the CA subject name hash value. It contains CA's
TLS CA Cert
(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.
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SIP Settings/NAT
Table 94: SIP Settings/NAT
Configure a static IP address and port (optional) used in outbound SIP messages
External Host
if the UCM6200 is behind NAT. If it is a host name, it will only be looked up once.
Use IP address in SDP
External TCP Port
External TLS Port
If enabled, the SDP connection will use the IP address resolved from the external
host.
Configure the externally mapped TCP port when the UCM6200 is behind a static
NAT or PAT.
Configures the externally mapped TLS port when UCM6200 is behind a static
NAT or PAT.
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
Local Network
will not be set correctly.
Address
A sample configuration could be as follows:
192.168.0.0/16
SIP Settings/TOS
Table 95: 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
Max
Registration/Subscrip
tion Time
Min
Registration/Subscrip
tion Time
Enable Relaxed DTMF
Configure the default duration (in seconds) of incoming/outgoing registration.
The default setting is 120.
Configure the maximum duration (in seconds) of incoming registration and
subscription allowed by the UCM6200. The default setting is 3600.
Configure the minimum duration (in seconds) of incoming registration and
subscription allowed by the UCM6200. The default setting is 60.
Select to enable relaxed DTMF handling. The default setting is "No".
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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
DTMF Mode
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".
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.
RTP Timeout
Note:
This setting doesn't apply to calls on hold.
When the call is on hold, if there is no RTP activity within the timeout (in seconds),
RTP Hold Timeout
the call will be terminated. This value of RTP Hold Timeout should be larger than
RTP Timeout. The default setting is no timeout.
This feature can be used to avoid abnormal call drop when the remote provider
requires RTP traffic during proceeding.
For example, when the call goes into voicemail and there is no RTP traffic sent
RTP Keep-alive
out from UCM, configuring this option can avoid voicemail drop.
When configured, RTP keep-alive packet will be sent to remote party at the
configured interval. If set to 0, RTP keep-alive is disabled.
100rel
Configure the 100rel setting on UCM6200. The default setting is “Yes”.
Trust Remote Party ID
Send Remote Party ID
Configure whether the Remote-Party-ID should be trusted. The default setting is
"No".
Configure whether the Remote-Party-ID should be sent or not. The default
setting is "No".
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.
Generate In-Band
•
Ringing
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 inband 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
Send Compact SIP
Headers
Configure the user agent string for the UCM6200.
If enabled, compact SIP headers will be sent. The default setting is "No".
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SIP Settings/SIP Trunk Prompt Tone
SIP Trunk Prompt Tone tab has been added to the UCM to help user choose which prompt will be played by
the UCM during call failure, the following voice message responses have been added and can be set to play
for 4XX, 5XX, and 6XX call failures:
•
Default for 404 and 604 status codes: “Your call can’t be completed as dialed. Please check the number
and dial again.”
•
Default for 5xx status codes: “Server error. Please check your device.”
•
Default for 403 and 603 status codes: “The call was rejected by the server. Please try again later.”
•
Default for all other status codes: “All circuits are busy now. Please try again later.”
Additionally, custom voice messages recorded and uploaded in PBX SettingsVoice PromptCustom
Prompt can be used for these failure responses instead of the default messages.
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Figure 187: SIP Trunk Prompt Tone
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Transparent Call-Info header
UCM supports transparent call info header in order to integrate GDS door system with GXP21XX Color phones,
the UCM will forward the call-info header to the phone in order to request the live view from GDS door system
and give the option to open the door via softkey.
Figure 188: Transparent Call-Info
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INTERFACE SETTINGS
Analog Hardware
The analog hardware (FXS port and FXO port) on the UCM6200 will be listed in this page. Click on
signaling preference for FXS port or configure ACIM settings for FXO port.
to edit
Select "Loop Start" or "Kewl Start" for each FXS port. And then click on "Update" to save the change.
Figure 189: FXS Ports Signaling Preference
For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for each
port. Or users could click on "Detect" and choose the detection algorithm, two algorithms exists (ERL, Pr) for the
UCM6200 to automatically detect the ACIM value. The detecting value will be automatically filled into the settings.
Figure 190: FXO Ports ACIM Settings
Table 96: PBX Interface Settings
Select country to set the default tones for dial tone, busy tone, ring tone and
Tone Region
etc to be sent from the FXS port. The default setting is "United States of
America (USA)".
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Advanced Settings
Select country to set the On-Hook Speed, Ringer Impedance, Ringer
Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum
FXO Opermode
Operational Loop Current, and AC Impedance as predefined for your country's
analog line characteristics. The default setting is "United States of America
(USA)".
Select country to set the On-Hook Speed, Ringer Impedance, Ringer
Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum
FXS Opermode
Operational Loop Current, and AC Impedance as predefined for your country's
analog line characteristics. The default setting is "United States of America
(USA)".
Configure to enable or disable override Two-Wire Impedance Synthesis
(TISS). The default setting is No.
FXS TISS Override
If enabled, users can select the impedance value for Two-Wire Impedance
Synthesis (TISS) override. The default setting is 600Ω.
Select the codec to be used for analog lines. North American users should
choose PCMU. All other countries, unless already known, should be assumed
PCMA Override
to be PCMA. The default setting is PCMU.
Note:
This option requires system reboot to take effect.
Configure whether normal ringing voltage (40V) or maximum ringing voltage
Boost Ringer
(89V) for analog phones attached to the FXS port is required. The default
setting is "Normal".
Fast Ringer
Low Power
Configure to increase the ringing speed to 25HZ. This option can be used with
"Low Power" option. The default setting is "Normal".
Configure the peak voltage up to 50V during "Fast Ringer" operation. This
option is used with "Fast Ringer". The default setting is "Normal".
If set to "Full Wave", false ring detection will be prevented for lines where Caller
Ring Detect
ID is sent before the first ring and proceeded by a polarity reversal, as in UK.
The default setting is "Standard".
Configure the type of Message Waiting Indicator on FXS lines. The default
FXS MWI Mode
setting is "FSK".
•
FSK: Frequency Shift Key Indicator
•
NEON: Light Neon Bulb Indicator.
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DAHDI Settings
When users encounter issues such as audio delay in outbound calls using the analog trunk, they can adjust
DAHDI settings on the UCM to attempt to lessen or resolve the issues.
Figure 191: DAHDI Settings
For the value of the option such as “32, half”:
The number in the option indicates the number of read/write buffers for TDM (DAHDI).
The “Half”, “Immediate” or “Full” option indicates the strategy when reading/writing data from buffer.
-
“Half”: Data will be read/written from buffer when half of the buffer is occupied with data.
-
“Immediate”: Read/write from buffer whenever there is data occupying the buffer.
-
“Full”: Data will be read/written from buffer when buffer is fully occupied with data.
Normally, DAHDI settings should be kept default and should be adjusted only when users encounter analog
trunk/Fax-related issues.
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CTI SERVER
UCM does support CTI server capabilities which are designed to be a part of the CTI solution suite provided by
Grandstream, including GXP21XX and GXP17XX enterprise IP phones along with GS Affinity app.
Mainly the UCM will by default listening on port TCP 8888 for the connections from GS affinity application in
order to interact, modify and serve data requests by the application which includes setting call features for the
connected extension as call forward and DND.
Users can change the listening port under the menu page, Web GUIValue-added FeaturesCTI Server as
shown on below screenshot:
Figure 192: CTI Server Listening port
More information about GS affinity and CTI Support on Grandstream products series please refer to the following
link: http://www.grandstream.com/sites/default/files/Resources/GS_Affinity_Guide.pdf
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ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)
The 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Value-added Features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/UCM_series_AMI_guide.pdf
----------------------------------------------------------------------------------------------------------------------------- ---------------------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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CRM INTEGRATION
Customer relationship management (CRM) is a term that refers to practices, strategies and technologies that
companies use to manage and analyze customer interactions and data throughout the customer lifecycle, with
the goal of improving business relationships with customers.
The UCM6200 series support two CRM API, SugarCRM and Salesforce CRM, which allows users to look for
contact information in the Contacts, Leads and / or Accounts tables, shows the contact record in CRM page, and
saves the call information in the contact’s history.
SugarCRM
Configuration page of the SugarCRM can be accessed via admin login, on the UCM Web GUIValue-added
FeaturesCRM.
Figure 193: SugarCRM Basic Settings
1. Select “SugarCRM” from the CRM System Dropdown in order to use SugarCRM.
Table 97: SugarCRM Settings
Select a CRM system from the Drop down, two CRM systems are
CRM System
available: Salesforce and SugarCRM.
CRM Server Address
Enter the IP address of the CRM server.
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Add the new number to this module if it can’t be found in the
Add Unknown Number
selected module.
Select from the “Available” list of lookups and press
Contact Lookups
to
select where the UCM can perform the lookups on the CRM tables,
Leads, Accounts, and Contacts.
Once settings on admin access are configured:
2. Click on
and
.
3. Logout from admin access.
4. Login to the UCM as user and navigate under “User PortalValue-added FeatureCRM User Settings”.
Click on “Enable CRM” and enter the username/password associated with the CRM account then click on
and
. The status will change from “Logged Out” to “Logged In”.
User can start then using SugarCRM features.
Figure 194: CRM User Settings
Salesforce CRM
Configuration page of the Salesforce CRM can be accessed via admin login, on the UCM webGUI under “Valueadded FeaturesCRM”.
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Figure 195: Salesforce Basic Settings
1. Select “Salesforce” from the CRM System Dropdown in order to use Salesforce CRM.
Table 98: Salesforce Settings
Select a CRM system from the Drop down, two CRM systems are
CRM System
available: Salesforce and SugarCRM.
Add the new number to this module if it can’t be found in the
Add Unknown Number
selected module.
Select from the “Available” list of lookups and press
to
select where the UCM can perform the lookups on the CRM
Contact Lookups
tables, Leads, Accounts, and Contacts.
Once settings on admin access are configured:
2. Click on
and
.
3. Logout from admin access.
4. Login to the UCM as user and navigate under “User PortalValue-added FeatureCRM User Settings”.
Click on “Enable CRM” and enter the username, password and Security Token associated with the CRM
account then click on
and
. The status will change from “Logged Out” to “Logged In”.
User can start then using Salesforce CRM features.
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Figure 196: Salesforce User Settings
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PMS INTEGRATION
UCM6200 series support Hotel Property Management System PMS, including check-in/check-out services,
wakeup calls, room status, Do Not Disturb which provide an ease of management for hotel applications. This
feature can be found on Web GUIValue-added FeaturesPMS.
Note: The PMS integration on UCM is currently supported only with one of the two following solutions.
HMobile PMS Connector
In this mode, the system can be divided into three parts:
•
PMS (Property Management System)
•
PMSI (Property Management System Interface)
•
PBX
Grandstream UCM6XXX series have integrated HMobile Connect PMSI which supports a large variety of PMS
software providing following hospitality features: Check-in, Check-out, set Room Status, Wake-up call and more.
The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and
PMS software, which is done through a middleware system (HMobile Connect) acting as interface between both
parties.
PMS Software
PMSI Middleware
(HMobile Connect)
Grandstream UCM
Figure 197: UCM & PMS interaction
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Mitel PMS
In this mode, the system can be divided into two parts:
•
PMS (Property Management System)
•
PBX
Grandstream UCM6XXX series have integrated Mitel PMS providing following hospitality features: Check-in,
Check-out, set Room Status, Wake-up call and more.
The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and
PMS software (Mitel). The communication between both parties is direct with no middleware.
Mitel PMS Software
Grandstream UCM
Figure 198: UCM & PMS interaction
The PMS module built-in the UCM supports the following features based on each solution:
Table 99: PMS Supported Features
Feature
Mitel
HMobile
Check-In
✓
✓
Check-out
✓
✓
Wake-up Call
✓
✓
Name Change
✓
✗
Update
✗
✓
Set Credit
✓
✗
Set Station Restriction
✓
✗
Room Status
✗
✓
Room Move
✗
✓
Do Not Disturb
✗
✓
Mini Bar
✗
✓
MSG
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Configuration
To use all PMS features please activate the feature code associated under “Call FeaturesFeature Codes”
-
Enable PMS
-
Update PMS Room Status
-
PMS Wake Up Service Activate
-
PMS Wake Up Service Deactivate
Basic Settings
On the UCM WebGUIValue-added FeaturesPMSBasic Settings” set the connection information for the
HMobile platform.
Table 100: PMS Basic Settings
Field
Description
Users can select the desired PMS module from the drop-down list.
PMS Module
Hmobile.
Mitel.
Prompt used when answering the wakeup calls it can be customized from
Wake Up Prompt
“PBX SettingsVoice PromptCustom Prompt.
PMS URL
Enter the PMS system URL
UCM Port
Enter the Port used by the PMS system
Username
Enter the Username to connect to the PMS system
Password
Enter the password to connect to the PMS system
PMS Features
Room Status
User can create Rooms by clicking on
, the following Figure will be displayed then.
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Figure 199: Create New Room
Click “Save” to create the new room, the fields above can be configured from the HMobile platform. Once set,
the following screen will be shown:
Figure 200: Room Status
User can create a batch of rooms as well by clicking on
, the following window will pop up:
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Figure 201: Add batch rooms
Wake Up Service
To create a New Wake up service, user can click on
, the following window will pop up:
Figure 202: Create New Wake Up Service
Table 101: PMS Wake up Service
Field
Description
Room Number
Select the room number where to call
Time
Set the time of the wakeup call
Action Status
Show the status of the call:
Type
•
Programmed: the call is scheduled for the time set
•
Cancelled: the call is canceled
•
Executed: the wakeup call is made
•
Single: The call will be made once on the specific time.
•
Daily: The call will be repeated every day on the specific time
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Once the call is made on the time specified, the following figure show the status of the wakeup call.
Figure 203: Wakeup Call executed
This call has been executed but has been rejected, that why we can see the “Busy” status.
Mini Bar
To
create
a
new
mini
bar,
click
on
under
UCM
webGUIValue-added
FeaturesPMSMini Bar, the following window will pop up:
Figure 204: Create New Mini Bar
Table 102: Create New Mini Bar
Code
Enter a non-existing extension number to be dialed when
using the mini bar feature.
Name
Enter a name for the mini bar.
Prompt
Select the Prompt to play once connected to the mini bar.
Skip Maid and Password
If enabled, the default maid code will be 0000, no
Authentication
authentication is required. (Enter 0000 followed by # to access
the consumer goods)
Enable Continuous Multi Goods
Billing
If enabled, please separate the goods’ codes by*.
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To create a new maid, click on
under UCM webGUIPBXPMSMini Bar, the following
window will pop up.
Figure 205: Create New Maid
Table 103: Create New Maid
Maid Code
Enter the Code to use when the maid wants to use the Mini Bar.
Secret
Enter the password associated with the maid.
To
create
a
new
consumer
goods,
click
on
under
webGUIPBXPMSMini Bar, the following window will pop up.
Figure 206: Create New Consumer Goods
Code
Enter the Goods Code.
Name
Enter the Name of the Goods
Success Prompt
Select the success prompt when typing the code of the goods by the
maid.
Failure Prompt
Select the failure prompt.
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UCM
The Minibar page displays as:
Figure 207: Mini Bar
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WAKEUP SERVICE
The WakeUp service can be used to schedule a reminder or wake up calls to any valid destination. This service
is available on the UCM6200 as a separated module.
There are three ways to set up Wakeup Service:
•
Using admin login
•
Using user portal
•
Using feature code
WakeUp Service using Admin Login
1. Login to the UCM as admin.
2. WakeUp service can be found under Web GUIValue-added FeaturesWakeup Service, click
on
to create a new wakeup service. The following window will pop up.
Figure 208: Create New Wakeup Service
3. Click
and
to apply the changes.
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Table 104: Wakeup Service
Enable Wakeup Service
Enable Wakeup service.
Name
Enter a name to identify the wakeup service.
Prompt
Select the prompt to play for that extension.
If disabled, users can select a specific date and time.
Custom Date
If enabled users can select multiple days of the week to perform the
wakeup.
Date
Select the date or dates when to performs the wakeup call.
Time
Select the time when to play the wakeup call.
Members
Select the members involved within the wakeup service group.
WakeUp Service from User Portal
1. Login to the user portal on the UCM6200.
2. WakeUp service can be found under “Value-added FeaturesWakeup Service”, click on
to create a new wakeup service.
3. Configures the Name, Prompt, Date and Time for the user to make the wakeup to.
4. Click
and
to apply the changes.
WakeUp Service using Feature Code
1. Login to the UCM as admin.
2. Enable “Wakeup Service” from the WebGUI under “Call FeaturesFeature Codes”.
3. Click
and
to apply the changes.
4. Dial “*36” which is the feature code by default to access to the UCM wakeup service to add, update,
activate or deactivate UCM wakeup service.
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ANNOUNCEMENTS CENTER
The 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.
Figure 209: Announcements Center
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Announcements Center Settings
Table 105: Announcements Center Settings
Configure a name for the newly created Announcements Center to identify this
Name
announcement center.
Enter a code number for the custom prompt. This code will be used in
combination with the group number. For example, if the code is 55, and group
number is 666. The user can dial 55666 to send prompt 55 to all members in
Code
group 666.
Note:
The combination number must not conflict with any number in the system such
as extension number or conference number.
This option is to set a custom prompt as an announcement to notify group
Custom Prompt
members. The file can be uploaded from page ‘Custom Prompt’. Click ‘Prompt’
to add additional record.
Configure the ring timeout for the group members. The default value is 30
Ring Timeout
seconds.
Group Settings
Table 106: Group Settings
Name
Configure a name for the newly created group to identify the group.
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
Number
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 GUI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.
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Figure 210: Announcements Center Group Configuration
In this example, group “Test” has number 666. Extension 1000, 1001 and 1002 are in this group.
1. Click
to create a new Announcement Center.
2. Give a name to the newly created Announcement Center.
3. Specify the code which will be used with group number to send the voice message to.
4. Select the message that will be used by the code from the Custom Prompt drop down menu. To create a
new Prompt, please click “Prompt” link and follow the instructions in that page.
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Figure 211: Announcements Center Code Configuration
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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 212: 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System
StatusDashboard.
Figure 213: StatusPBX Status
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Trunks
Users could see all the configured trunk status in this section.
Figure 214: Trunk Status
Table 107: Trunk Status
Display trunk status.
•
Analog trunk status:
Available
Busy
Unavailable
Unknown Error
Status
•
SIP Peer trunk status:
Unreachable: The hostname cannot be reached.
Unmonitored: Heartbeat feature is not turned on to be monitored.
Reachable: The hostname can be reached.
•
SIP Register trunk status:
Registered
Unrecognized Trunk
Trunks
Display trunk name
Display trunk Type:
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.
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Extensions
Extensions Status can be seen from the same configuration page, users can go under Web
GUIExtension/TrunkExtensions and following page will be displayed listing the extensions and their status
information.
Figure 215: Extension Status
Table 108: Extension Status
Display extension number (including feature code). The color indicator has the
following definitions.
Status
•
Green: Free
•
Blue: Ringing
•
Yellow: In Use
•
Grey: Unavailable
Presence Status
Display the presence status of the extension.
Extension
Display the extension number.
CallerID Name
First name and last name of the extension.
IP and Port
Display the IP and port number of the registered device.
Display Email Notification status for the extension.
Email
When notification is waiting for be sent, shows
and once sent it will display
Displays extension type.
Terminal Type
•
SIP User
•
IAX User
•
Analog User
•
Ring Groups
•
Voicemail Groups
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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 216: UCM6204 Interfaces Status
Table 109: Interface Status Indicators
USB connected.
USB disconnected.
SD Card connected.
SD Card disconnected.
LAN/WAN connected.
LAN/WAN not configured.
LAN/WAN disconnected.
FXS/FXO connected.
FXS/FXO waiting.
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FXS/FXO busy.
FXS/FXO not configured.
FXS/FXO disconnected.
System Status
The UCM6200 system status can be accessed via Web GUIStatusSystem Status, which displays the
following system information.
General
Under Web GUISystem StatusSystem InformationGeneral, users could check the hardware and
software information for the UCM6200. Please see details in the following table.
Table 110: System StatusGeneral
System Status System Information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.
Recovery
Recovery version.
Network
Under Web GUISystem StatusSystem InformationNetwork, users could check the network information
for the UCM6200. Please see details in the following table.
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Table 111: System StatusNetwork
System 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 GUISystem StatusDashboardStorage
Usage. It shows the available and used space for Space Usage and Inode Usage.
Space Usage includes:
•
Configuration partition
This partition contains PBX system configuration files and service configuration files.
•
Data partition
Voicemail, recording files, IVR file, Music on Hold files etc.
•
USB disk
USB disk will display if connected.
•
SD Card
SD Card will display if connected.
Inode Usage includes:
•
Configuration partition
•
Data partition
Note:
Inode is the pointer used for file reference in the system. The system usually has limited resources of pointers
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Figure 217: 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
GUISystem
StatusDashboardResource Usage, the current CPU usage and Memory usage are shown in the pie chart.
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Figure 218: 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MaintenanceSystem Events. The following event
are currently supported on the UCM6200 which will have alert and/or Email generated if occurred:
Disk Usage
External 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
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Figure 219: 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. External Disk Usage
Figure 220: System EventsAlert Events Lists: External Disk Usage
•
Detect Cycle: The UCM6200 will perform the External disk usage detection based on this cycle. Users
can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
•
Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM6200 system will
send the alert.
3. Memory Usage
Figure 221: 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.
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4. System Crash
Figure 222: System EventsAlert Events Lists: System Crash
•
Detect Cycle: The UCM will detect the event at each cycle based on the specified time. Users can
enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
Click on the switch
to turn on/off the alert and Email notification for the event. Users
could also select the checkbox for each event and then click on button "Alert On", "Alert Off", "Email
Notification On", "Email Notification Off" to control the alert and Email notification configuration.
Alert Log
Under Web GUIMaintenanceSystem EventsAlert Log, system messages from triggered system events
are listed as alert logs. The following screenshot shows system crash alert logs.
Figure 223: 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:
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1. Generate Alert: Generated when alert events happen, for example, alert logs for disk usage exceeding the
alert threshold.
2. Restore to Normal: Generated when alert events being cleared, for example, logs for disk usage dropping
back below the alert threshold.
User could filter out alert logs of “Generate Alert” or “Restore to Normal” by specifying the type according to
need. The following figure shows an example of filtering out alert logs of type of “Restore to Normal”.
Figure 224: Filter for Alert Log
Alert Contact
Users could add administrator's Email address under Web GUIMaintenanceSystem EventsAlert
Contact to send the alert notification to. Up to 10 Email addresses can be added.
CDR
CDR (Call Detail Record) is a data record generated by the PBX that contains attributes specific to a single
instance of phone call handled by the PBX. It has several data fields to provide detailed description for the call,
such as phone number of the calling party, phone number of the receiving party, start time, call duration, etc.
On the UCM6200, the CDR can be accessed under Web GUICDRCDR. Users could filter the call report by
specifying the date range and criteria, depending on how the users would like to include the logs to the report.
Click on "Search" button to display the generated report.
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Figure 225: CDR Filter
Table 112: CDR Filter Criteria
Call Type
Groups the following:
•
Inbound calls: Inbound calls are calls originated from a non-internal source
(like a VoIP trunk) and sent to an internal extension.
•
Outbound calls: Outbound calls are calls sent to a non-internal source (like
a VoIP trunk) from an internal extension.
•
Internal calls: Internal calls are calls from one internal extension to another
extension, which are not sent over a trunk.
•
External calls: External calls are calls sent from one trunk to another trunk,
which are not sent to any internal extension.
Status
Filter with the call status, the available statuses are the following:
•
Answered
•
No Answer
•
Busy
•
Failed
Source Trunk
Select source trunk(s) and the CDR of calls going through inbound the trunk(s) will be
Name
filtered out.
Destination Trunk
Select destination trunk(s) and the CDR of calls going outbound through the trunk(s)
Name
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Action Type
Account Code
Filter calls using the Action Type, the following actions are available:
•
Dial
•
Announcements
•
Callback
•
Call Forward
•
Conference
•
Disa
•
Fax
•
Follow Me
•
IVR
•
Page
•
Parked Call
•
Queue
•
Ring Group
•
Transfer
•
VFax
•
VM
•
VMG
•
Wakeup
Select the account Code to filter with. If pin group CDR is enabled, the call with pin
group information will be displayed as part of the CDR under Account Code Field.
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, caseinsensitive, repeatable, only appears in the end.
For example:
3XXX: It will filter out CDR that having caller number with leading digit 3 and of 4 digits’
length.
3.: It will filter out CDR that having caller number with leading digit 3 and of any length.
Caller Name
Enter the caller name to filter the CDR report. CDR with the matching caller name will
be filtered out.
Callee Number
Enter the callee number to filter the CDR report. CDR with the matching callee number
will be filtered out.
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The call report will display as the following figure shows.
Figure 226: Call Report
The CDR report has the following data fields:
•
Start Time
Format: 2016-09-03 00:06:16
•
Call Type
Example:
IVR
DIAL
WAKEUP
•
Call From
Example format:
"John Doe" 2000
•
Call To
Example format:
2002
•
Call Time
Format: 0:00:02
•
Talk Time
Format: 0:00:00
•
Account Code
Example format:
Grandstream/Test
•
Status
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Answered, Busy, No answer or Failed.
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Users could perform the following operations on the call report.
•
Sort by “Start Time”
Click on the header of the column to sort the report by "Start Time". Clicking on "Start Time" again will
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 227: Call Report Entry with Audio Recording File
•
Automatic Download CDR Records
User could configure the UCM6200 to automatically download the CDR records and send the records to
multiple Email recipients in a specific hour. Click on “Automatic Download Settings”, and configure the
parameters in the dialog below.
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Figure 228: Automatic Download Settings
To receive CDR record automatically from Email, check “Enable” and select a time period “By Day” “By Week”
or “By Month”, select Hour of the day as well for the automatic download period. Make sure you have entered
an Email or multiple email addresses where to receive the CDR records.
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
GUI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 229: CDR Report
Figure 230: Detailed CDR Information
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Downloaded CDR File
The downloaded CDR (.csv file) has different format from the Web GUI CDR. Here are some descriptions.
•
Caller number, Callee number
"Caller number": the caller ID.
"Callee number": the callee ID.
If the "Source Channel" contains "DAHDI", this means the call is from FXO/PSTN line.
Figure 231: Downloaded CDR File Sample
•
Context
There are different context values that might show up in the downloaded CDR file. The actual value can vary
case by case. Here are some sample values and their descriptions.
from-internal: internal extension makes outbound calls.
ext-did-XXXXX: inbound calls. It starts with "ext-did", and "XXXXX" content varies case by case, which also
relate to the order when the trunk is created.
ext-local: internal calls between local extensions.
•
Source Channel, Dest Channel
Sample 1:
Figure 232: 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 233: Downloaded CDR File Sample - Source Channel and Dest Channel 2
"SIP" means it's a SIP call. There are three possible format:
(a) PJSIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string
and can be ignored.
(c) PJSIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out through
the trunk.
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(c) PJSIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this trunk.
The last XXXXX is a random string and can be ignored.
There are some other 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
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.
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Figure 234: CDR Statistics
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Table 113: CDR Statistics Filter Criteria
Trunk Type
Select one of the following trunk type.
Call Type
•
All
•
SIP Calls
•
PSTN Calls
Select one or more in the following checkboxes.
Time Range
•
Inbound calls
•
Outbound calls
•
Internal calls
•
External calls
•
All calls
•
By month (of the selected year).
•
By week (of the selected year).
•
By day (of the specified month for the year).
•
By hour (of the specified date).
•
By range. For example, 2016-01 To 2016-03.
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.
Figure 235: 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
•
Click on
•
To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding column. Click
to download the recording file in .wav format.
to delete the recording file.
on the title again can switch the sorting mode between ascending order or descending order.
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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 114: 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.
Username
Configure the Username for API Authentication.
Password
Configure the Password for API Authentication.
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.
Permitted IPs
Specify a list of IP addresses permitted by API. This creates an AIP-specific access
control list. Multiple entries are allowed.
For example, "192.168.5.20/255.255.255.255" denies access from all IP addresses
except 192.168.5.20.
The default setting is blank, meaning all IPs will be denied. Users must set permitted
IP address before connecting to the API.
Reset Certificates
Press
button to restore UCM’s default certificates.
For more details on CDR API (Access to Call Detail Records) and REC API (Access to Call Recording Files),
please refer the document in the link here:
http://www.grandstream.com/sites/default/files/Resources/ucm6xxx_cdr_rec_api_guide.pdf
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USER PORTAL
Users could log into their web GUI portal using the extension number and user password. When an extension is
created in the UCM62XX, the corresponding user account for the extension is automatically created. The user
portal allows access to a variety of features which include user information, extension configuration and CDR as
well as settings and managing value-added features like WebRTC, Fax Sending, Call Queue, Wakeup Service
and CRM.
Users also can access their personal data files (call recordings, Fax files, Voicemail Prompts …).
The login credentials are configured by Super Admin. The following figure shows the dialog of editing the account
information by Super Admin. The User Name must be the extension number and it’s not configurable, and the
password is set on “User Password” field and it should not be confused with the SIP extension password.
Figure 236: Edit User Information by Super Admin
The following screenshot shows an example of login page using extension number 1000 as the username.
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Figure 237: User Portal Login
After login, the Web GUI display is shown as below.
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Figure 238: User Portal Layout
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After successful login, the user has the following three configuration tabs:
Basic Information
Under this menu, the user can configure and change his/her personal information including (first name, last
name, password, email address, department…). And they can also set and activate their extension features
(presence status, call forward, DND ….) to be reflected on the UCM.
Also, the user can see from this menu the Call Details Records and search for specific ones along with the
possibility to download the records on CSV format for later usage.
Personal Data
Under this section, the user can access and manage their personal data files which includes (voicemail files, call
recordings, and fax files) along with the possibility to set Follow me feature to without requesting the Super admin
to set the feature from admin account.
Value-added Features
On this section, the user has access to manage and use all rich value-added features which includes.
+ WebRTC connection and making calls from the browser.
+ Sending Fax files using PDF or TIF/TIFF format.
+ If user is a member of call queue, they can check the queue’s activity from the “Call Queue” section.
+ Create and enable WakeUp service.
+ Enable and configure CRM connection to either SugarCRM or Salesforce.
For the configuration parameter information in each page, please refer to [Table 115: 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.
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MAINTENANCE
User Management
User management is on Web GUIMaintenance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 GUI using their
extension number and password. All existing user accounts for Web GUI login will be displayed on User
Management page as shown in the following figure.
Figure 239: User Management Page Display
User Information
When logged in as Super Admin, click on
to create a new account for Web GUI user. The
following dialog will prompt. Configure the parameters as shown in below table.
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Figure 240: Create New User
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Table 115: User ManagementCreate New User
Configure a username to identify the user which will be required in Web GUI login.
User Name
Letters, digits and underscore are allowed in the user name.
Configure a password for this user which will be required in Web GUI login. Letters,
User Password
digits and underscore are allowed.
This is the role of the Web GUI user. Currently only “Admin” is supported when
Privilege
Super Admin creates a new user.
Department
Fax
Email Address
First Name
Enter the necessary information to keep a record for this user.
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 241: User Management – New Users
Custom Privilege
Four privilege levels are supported:
•
Super Administrator
-
This is the highest privilege. Super Admin can access all pages on 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.
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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 GUIMaintenanceChange Information
-
page.
Super Admin could view operations done by all the users in Web GUIMaintenanceUser
-
ManagementOperation Log
•
Administrator
-
Users with “Admin” privilege can only be created by “Super Admin” user.
-
“Admin” privilege users are not allowed to access the following pages:
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 GUI login is created automatically by the system when a new extension is
created.
-
The user could log in the Web GUI with the extension number and password to access user information,
extension configuration and CDR of that extension.
-
The SuperAdmin user can click on
on the “General_User” in order to enable/disable the custom
privilege from deleting their own recording files, changing SIP credentials and disabling voicemail
service in their user portal account.
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Figure 242: General User
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•
Custom Privilege
The Super Admin user can create users with different privileges. 6 modules are available for privilege
customization.
-
System Status
-
Conference
-
System Events
-
CDR Records
-
CDR API
-
Wakeup Service
Figure 243: Create New Custom Privilege
Log in UCM6200 as super admin and go to MaintenanceUser ManagementCustom Privilege, create
privilege with customized available modules.
To assign custom
privilege to a sub-admin, navigate to UCM Web GUIMaintenanceUser
ManagementUser InformationCreate New User/Edit Users, select the custom privilege from “Privilege”
option.
Concurrent Multi-User Login
When there are multiple Web GUI users created, concurrent multi-user login is supported on the 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.
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Figure 244: Multiple User Operation Error Prompt
Change Password
After logging in the UCM6200 Web GUI 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 GUI access password.
1. Go to Web GUIMaintenanceChange Information page.
2. Enter the old password first.
3. Enter the new password and re-type the new password to confirm. The new password has to be at least 4
characters. The maximum length of the password is 30 characters.
4. Configure the Email Address that is used when login credential is lost.
5. Click on "Save" and the user will be automatically logged out.
6. Once the web page comes back to the login page again, enter the username "admin" and the new password
to login.
Figure 245 : 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
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Change binding 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 GUISystem SettingsUser
ManagementChange Binding Email.
Figure 246: Change Binding Email
Table 116: Change Binding Email option
Enter the password
of the account
Email Address
Enter the current login user credential for UCM6200
Email Address is used to retrieve password when password is lost
Login Settings
After the user logs in the UCM6100 Web GUI, the user will be automatically logged out after certain timeout, or
he/she can be banned for a specific period if the login timeout is exceeded. Those values can be specified under
UCM6100 web GUIMaintenanceChange InformationLogin 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 GUI within the timeout, the user will be logged out automatically. After that, the Web GUI will
be redirected to the login page and the user will need to enter username and password to log in.
If set to 0, there is no timeout for the Web GUI login session and the user will not be automatically logged out.
“Maximum number of login attempts” can prevent the UCM6100 from brutal force decryption, if this number
is exceeded user IP address will be banned from accessing the UCM for a period of time based on user
configuration, the default value is 5.
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“User ban period” specify the period of time in minutes an IP will be banned from accessing the UCM if the User
max number of try login is exceeded, the default value is 5.
“Login Banned User List” show the list of IPs’ banned from the UCM.
“Login White List” User can add a list of IPs’ to avoid the above restriction, thus, they can exceed the User max
number of try login.
Figure 247: Login Timeout Settings
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 GUI users, for example,
Web GUI login, creating trunk, creating outbound rule and etc. There are 7 columns to record the operation
details “Date”, “User Name”, “IP Address”, “Results”, “Page Operation”, “Specific Operation” and “Remark”.
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Figure 248: 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 117: 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.
The page where the operation is made. For example, login, logout, delete user, create
Page Operation
trunk and etc.
Specific Operation
Click on
Remark
Allows users to add notes and remarks to each operation
to view the options and values configured by this operation.
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User could also filter the operation logs by time condition, IP address and/or username. Configure these
conditions and then click on
.
Figure 249: 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
filtered result of operation logs. Or users can click on
to delete the
to delete all operation logs at once.
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;
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Examples of valid URLs:
firmware.grandstream.com/BETA
The upgrading configuration can be accessed via Web GUIMaintenanceUpgrade.
Figure 250: Network Upgrade
Table 118: 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.
1. Enter the firmware server path under Web GUIMaintenanceUpgrade.
2. Click on "Save". Then reboot the device to start the upgrading process.
3. Please be patient during upgrading process. Once done, a reboot message will be displayed in the LCD.
4. 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.
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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.
1. Download the latest UCM6200 firmware file from the following link and save it in your PC.
http://www.grandstream.com/support/firmware
2. Log in the Web GUI as administrator in the PC.
3. Go to Web GUIMaintenanceUpgrade, upload the firmware file by clicking on “choose file to upload”
and select the firmware file from your PC. The default firmware file name is ucm6200fw.bin
Figure 251: Local Upgrade
Figure 252: Upgrading Firmware Files
4. Wait until the upgrading process is successful and a window will be popped up in the Web GUI.
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Figure 253: Reboot UCM6200
5. Click on "OK" to reboot the UCM6200 and check the firmware version after it boots up.
----------------------------------------------------------------------------------------------------------------------------- ---------------------Notes:
-
Please do not interrupt or power cycle the UCM6200 during upgrading process.
-
The firmware file name allows the use of the following special characters: “_@#*~&”.
----------------------------------------------------------------------------------------------------------------------------- ----------------------
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;
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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
GUIMaintenanceBackupBackup/Restore.
Click on
to create a new backup file. Then the following dialog will show.
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Figure 254: Create New Backup
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Web
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
Click on
, or delete
it from the UCM6200 internal storage or the external device.
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.
It is also possible to load backup files from UCM6100 to UCM6200 and vice versa.
Please make sure the FXO port settings, total number of extensions and total number of conference rooms are
compactable before restoring to another UCM model. Otherwise it will prompt a warning and stop the restore
process as shown below:
Figure 255: Restore Warning
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Figure 256: Backup / Restore
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The
option allows UCM to perform automatically backup on the user specified time.
Regular
backup file can only be stored in USB / SD card / SFTP server. User is allowed to set backup time from 0-23 and
how frequent the backup will be performed.
Figure 257: Local Backup
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.
The client account supports special characters such as @ or “.” Allowing the use email address as SFTP
accounts. It allows users as well to specify the destination directory on SFTP server for backup file. If the directory
doesn’t exist on the destination, UCM6200 will create the directory automatically
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Figure 258: Data Sync
Table 119: 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.
Destination Directory
Specify the directory in SFTP server to keep the backup file. Format:
‘xxx/xxx/xxx’, If this directory does not exist, UCM will create this directory
automatically.
Sync Time
Enter 0-23 to specify the backup hour of the day.
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Before saving the configuration, users could click on
. 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 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Backup/Restore.
•
A list of previous configuration backups is displayed on the web page. Users could click on
of the
desired backup file and it will be restored to the 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 259: Restore UCM6200 from Backup File
----------------------------------------------------------------------------------------------------------------------------- ---------------------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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System Cleanup/Reset
Reset and Reboot
Users could perform reset and reboot under Web GUIMaintenanceSystem Cleanup/Reset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 260: Reset and Reboot
Cleaner
Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under Web
GUIMaintenanceSystem Cleanup/ResetCleaner.
The following screenshot show the settings and parameters to configure the cleaner feature on UCM6200.
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Figure 261: Cleaner
Table 120: Cleaner Configuration
Enable CDR Cleaner
Enable the CDR Cleaner function.
Clean Files in External
If enabled the files in external device (USB/SD card) will be atomically cleaned
Device
up as configured.
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 Queue Log
Cleaner
Queue Log Clean Time
Enable scheduled queue log cleaning. By default, is disabled.
Enter the hour of the day to start the cleaning. The valid range is 0-23.
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Clean Interval
Enter how often (in days) to clean queue logs. The valid range is 1-30.
Enable File Cleaner
Enter the Voice Records Cleaner function.
Choose Cleaner File
Select the files for system automatic clean.
•
Basic Call Recording Files.
•
Conference Recording Files.
•
Call Queue Recording Files.
•
Voicemail Files.
•
Fax
•
Backup Files.
File Clean Threshold
Specify the threshold of local storage usage from 0 to 99 (in percentage).
File Clean Time
Enter 0-23 to specify the hour of the day to clean up the files.
File Clean Interval
Enter 1-30 to specify the day of the month to clean up the files.
All the cleaner logs will be listed on the bottom of the page.
USB/SD Card Files Cleanup
Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under Web
GUIMaintenanceSystem Cleanup/ResetUSB / SD Card Files Cleanup.
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Figure 262: USB/SD Card Files Cleanup
Table 121: USB/SD Card Files Cleanup
Current Path
Displays the current path.
Directory
Select the directory user want to clean.
Delete Selected File
Select multiple entries to delete from USB or SD card.
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 "notice" and "verb" levels
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besides "error" level.
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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 room.
----------------------------------------------------------------------------------------------------------------------------- ---------------------Note:
Syslog is usually for debugging and troubleshooting purpose. Turning on all levels for all syslog modules is not
recommended for daily usage. Too many syslog prints might cause traffic and affect system performance.
-------------------------------------------------------------------------------------------- -------------------------------------------------------
Network Troubleshooting
On the UCM6200, users could capture traces, ping remote host and traceroute remote host for troubleshooting
purpose under Web GUIMaintenanceNetwork Troubleshooting.
The following sections shows the steps to capture different types of traffic traces for analysis purposes.
Ethernet Capture
The captured trace can be downloaded for analysis. The instructions or result will be displayed in the Web GUI
output result.
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Figure 263: Ethernet Capture
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Table 122: Ethernet Capture
Interface Type
Select the network interface to monitor.
Enable SFTP Data Sync
Check this box to save the capture files in the SFTP server.
Please make sure the configuration of data synchronization works
before.
Storage to External Device
Check this box to activate storage of the capture either on the
USB or SD Card.
Capture Filter
Enter the filter to obtain the specific types of traffic, such as (host,
src, dst, net, proto…).
Start
Click to start the trace.
Stop
Click to stop the trace.
Download
Click to download the trace if trace is stored locally.
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 264: 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 265: Traceroute
Signaling Troubleshooting
Analog Record Trace
•
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Signal 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".
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7. Click on "Download" to download the analog record trace.
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Figure 266: Troubleshooting Analog Trunks
•
A key Dial-up FXO
Users can directly set a PSTN number on the “External Extension” text box to troubleshoot issues related to
the analog trunk easily, the following steps shows how to use this feature:
1. Configure analog trunk on UCM, including outbound route.
2. Enter a reachable external number in “External Extension”.
3. Press “Start” button. The call will be initiated to the external number.
4. Answer and finish the call before pressing “Stop” button.
The trace will be available for analysis to download after output result shows “Done! Click on Download to
download the captured packets”.
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Figure 267: A Key Dial-up FXO
Note: When using a Key Dial-up FXO feature the outbound trunk for the analog trunk need to have internal
permission. As well as it should be the trunk with the highest outbound route priority.
After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream
Technical support in the following link for further assistance if the issue is not resolved.
http://www.grandstream.com/index.php/support
Service Check
Enable Service Check to periodically check 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 268: Service Check
Network Status
In UCM6200 Web GUISystem Status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 269: Network Status
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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
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