Grandstream Networks, Inc.

UCM6200 Series IP PBX

User Manual
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©2021 Grandstream Networks, Inc. http://www.grandstream.com

All rights reserved. Information in this document is subject to change without notice. Reproduction or transmittal
of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express
written permission of Grandstream Networks, Inc. is not permitted.
The latest electronic version of this user manual is available for download here:
http://www.grandstream.com/support
Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc. in
the United States, Europe and other countries.

CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this product
in any way other than as detailed by this User Manual, could void your manufacturer warranty.

WARNING
Please do not use a different power adaptor with your devices as it may cause damage to the products and
void the manufacturer warranty.

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UCM6200 Series User Manual
Version 1.0.20.38
GNU GPL INFORMATION

UCM62xx 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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UCM6200 Series User Manual
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 Table of Content

DOCUMENT PURPOSE........................................................................... 27

CHANGE LOG ......................................................................................... 28

Firmware Version 1.0.20.38................................................................................................................ 28
Firmware Version 1.0.20.34................................................................................................................ 28
Firmware Version 1.0.20.31................................................................................................................ 28
Firmware Version 1.0.20.28................................................................................................................ 29
Firmware Version 1.0.20.23................................................................................................................ 29
Firmware Version 1.0.20.22................................................................................................................ 29
Firmware Version 1.0.20.20................................................................................................................ 29
Firmware Version 1.0.20.17................................................................................................................ 29
Firmware Version 1.0.19.29................................................................................................................ 34
Firmware Version 1.0.19.27................................................................................................................ 34
Firmware Version 1.0.19.21................................................................................................................ 35
Firmware Version 1.0.18.13................................................................................................................ 35
Firmware Version 1.0.18.12................................................................................................................ 35
Firmware Version 1.0.18.9.................................................................................................................. 35
Firmware Version 1.0.17.16................................................................................................................ 36
Firmware Version 1.0.16.20................................................................................................................ 37
Firmware Version 1.0.16.18................................................................................................................ 37
Firmware Version 1.0.15.16................................................................................................................ 38
Firmware Version 1.0.14.24................................................................................................................ 38
Firmware Version 1.0.14.23................................................................................................................ 38
Firmware Version 1.0.14.21................................................................................................................ 39
Firmware Version 1.0.13.14................................................................................................................ 39
Firmware Version 1.0.12.19................................................................................................................ 40
Firmware Version 1.0.11.27 ................................................................................................................ 41
Firmware Version 1.0.0.7 .................................................................................................................... 41

WELCOME ............................................................................................... 42

PRODUCT OVERVIEW ............................................................................ 43

Technical Specifications ..................................................................................................................... 43

INSTALLATION ........................................................................................ 46
Equipment Packaging ......................................................................................................................... 46
Connect Your UCM6200 ..................................................................................................................... 46

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 Connect the UCM6202 ............................................................................................................... 46
Connect the UCM6204 ............................................................................................................... 47
Connect the UCM6208 ............................................................................................................... 48
Safety Compliances ............................................................................................................................ 49
Warranty ............................................................................................................................................. 49

GETTING STARTED ................................................................................ 50

Use the LCD Menu ............................................................................................................................. 50
Use the LED Indicators ....................................................................................................................... 52
Using the Web UI ............................................................................................................................... 53
Accessing the Web UI ................................................................................................................. 53
Setup Wizard............................................................................................................................... 54
Main Settings .............................................................................................................................. 55
Web GUI Languages .................................................................................................................. 55
Web GUI Search Bar .................................................................................................................. 56
Saving and Applying Changes .................................................................................................... 56
Setting Up an Extension ..................................................................................................................... 57

SYSTEM SETTINGS ................................................................................ 58

HTTP Server ....................................................................................................................................... 58
Network Settings ................................................................................................................................ 59
Basic Settings ............................................................................................................................. 59
DHCP Client List ......................................................................................................................... 65
802.1X ......................................................................................................................................... 66
Static Routes ............................................................................................................................... 67
Port Forwarding .......................................................................................................................... 70
OpenVPN® ......................................................................................................................................... 72
DDNS Settings ................................................................................................................................... 73
Security Settings ................................................................................................................................. 76
Static Defense ............................................................................................................................. 76
Dynamic Defense ........................................................................................................................ 78
Fail2ban ...................................................................................................................................... 79
TLS Security................................................................................................................................ 81
SSH Access ................................................................................................................................ 82
LDAP Server ....................................................................................................................................... 82
LDAP Server Configurations ....................................................................................................... 83
LDAP Phonebook ........................................................................................................................ 84
LDAP Client Configurations ........................................................................................................ 87
Time settings ...................................................................................................................................... 91
Automatic Date and Time ........................................................................................................... 91
Set Time Manually ...................................................................................................................... 92

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 NTP Server ................................................................................................................................. 93
Office Time .................................................................................................................................. 93
Holiday ........................................................................................................................................ 94
Email Settings ..................................................................................................................................... 96
Email settings .............................................................................................................................. 96
Email Templates .......................................................................................................................... 99
Email Send Log ......................................................................................................................... 100
Recordings Storage .......................................................................................................................... 102

PROVISIONING ..................................................................................... 104

Overview ........................................................................................................................................... 104
Configuration Architecture for End Point Device .............................................................................. 104
Auto Provisioning Settings................................................................................................................ 105
Discovery .......................................................................................................................................... 108
Uploading Devices List ..................................................................................................................... 110
Managing discovered devices ........................................................................................................... 111
Global configuration .......................................................................................................................... 112
Global policy.............................................................................................................................. 112
Global Templates ...................................................................................................................... 120
Model configuration .......................................................................................................................... 122
Model templates ........................................................................................................................ 122
Model Update ............................................................................................................................ 124
Device Configuration ........................................................................................................................ 126
Create New Device ................................................................................................................... 126
Manage Devices ....................................................................................................................... 127
Example Application ......................................................................................................................... 133

EXTENSIONS......................................................................................... 138
Create New User .............................................................................................................................. 138
Create New SIP Extension ....................................................................................................... 138
Create New IAX Extension ....................................................................................................... 146
Create New FXS Extension ...................................................................................................... 150
Batch Add Extensions ....................................................................................................................... 155
Batch Add SIP Extensions ........................................................................................................ 155
Batch Add IAX Extensions ........................................................................................................ 158
Batch Extension Resetting Functionality .......................................................................................... 160
Search and Edit Extension ............................................................................................................... 160
Export Extensions ............................................................................................................................. 162
Import Extensions ............................................................................................................................. 162
Extension Details .............................................................................................................................. 169
E-mail Notification ............................................................................................................................. 170

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 Multiple Registrations per Extension ................................................................................................ 172
SMS Message Support ..................................................................................................................... 173

EXTENSION GROUPS........................................................................... 174

Configure Extension Groups ............................................................................................................ 174
Using Extension Groups ................................................................................................................... 175

ANALOG TRUNKS ................................................................................ 176

Analog Trunk Configuration .............................................................................................................. 176
PSTN Detection ................................................................................................................................ 180

VOIP TRUNKS ....................................................................................... 183

VoIP Trunk Configuration .................................................................................................................. 183
Trunk Groups .................................................................................................................................... 194
Direct Outward Dialing (DOD) .......................................................................................................... 195

SLA STATION ........................................................................................ 198

Create/Edit SLA Station .................................................................................................................... 198
Sample Configuration ....................................................................................................................... 199

CALL ROUTES ...................................................................................... 201

Outbound Routes ............................................................................................................................. 201
Configuring Outbound Routes .................................................................................................. 201
Outbound Blacklist .................................................................................................................... 204
PIN Groups ............................................................................................................................... 206
Inbound Routes ................................................................................................................................ 210
Inbound Rule Configurations .................................................................................................... 211
Inbound Route: Prepend Example ............................................................................................ 215
Inbound Route: Multiple Mode .................................................................................................. 216
Inbound Route: Route-Level Mode ........................................................................................... 218
Inbound Route: Inbound Mode BLF Monitoring ........................................................................ 219
Inbound Route: Import/Export Inbound Route .......................................................................... 220
FAX Detection ........................................................................................................................... 221
FAX with Two Media.................................................................................................................. 221
Blacklist Configurations ............................................................................................................. 221

CONFERENCE....................................................................................... 223
Conference Room Configurations .................................................................................................... 223
Conference Call Operations ............................................................................................................. 226
Join a Conference Call .............................................................................................................. 226
Invite Other Parties to Join Conference .................................................................................... 226

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 During The Conference ............................................................................................................. 227
Google Service Settings Support .............................................................................................. 229
Conference Schedule ....................................................................................................................... 231
Cleaner Options ........................................................................................................................ 234
Contact Group .................................................................................................................................. 235
Contact Group Configurations .................................................................................................. 236
Conference Recordings .................................................................................................................... 236
Conference Call Statistics ................................................................................................................ 237

VIDEO CONFERENCE........................................................................... 239

Video Conference Room Configurations .......................................................................................... 239
Conference Settings ......................................................................................................................... 240
Conference Schedule ....................................................................................................................... 241
Wave WebRTC Video Calling & Conferencing ................................................................................. 242

IPVIDEOTALK MEETINGS .................................................................... 245

IVR ......................................................................................................... 248

Configure IVR ................................................................................................................................... 248
IVR Black/Whitelist ........................................................................................................................... 251
Create Custom Prompt ..................................................................................................................... 253

VOICE PROMPT .................................................................................... 254

Language Settings ............................................................................................................................ 254
Download and Install Voice Prompt Package ........................................................................... 254
Upload Language Package....................................................................................................... 256
Custom Prompt ................................................................................................................................. 257
Record New Custom Prompt .................................................................................................... 257
Upload Custom Prompt ............................................................................................................. 257
Download All Custom Prompt ................................................................................................... 258
Username Prompt Customization .................................................................................................... 258
Upload Username Prompt File from Web GUI ......................................................................... 258
Record Username via Voicemail Menu ..................................................................................... 259

VOICEMAIL ............................................................................................ 260

Configure Voicemail .......................................................................................................................... 260
Access Voicemail .............................................................................................................................. 262
Leaving Voicemail ............................................................................................................................. 263
Voicemail Email Settings .................................................................................................................. 264
Configure Voicemail Group............................................................................................................... 265

RING GROUP......................................................................................... 266

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 Configure Ring Group ....................................................................................................................... 266
Remote Extension in Ring Group ..................................................................................................... 269

RESTRICT CALLS ................................................................................. 272

Configure Restrict Calls .................................................................................................................... 272

PAGING AND INTERCOM GROUP ....................................................... 274

Configure Paging/Intercom Group .................................................................................................... 274
Configure Multicast Paging ....................................................................................................... 274
Configure 2-way Intercom ......................................................................................................... 275
Configure 1-way Paging ............................................................................................................ 276
Configure Announcement Paging ............................................................................................. 277
Configure Private Intercom ....................................................................................................... 278
Paging/Intercom Group Settings ............................................................................................... 280
Configure a Scheduled Paging/Intercom .......................................................................................... 280

CALL QUEUE ........................................................................................ 282

Configure Call Queue ....................................................................................................................... 282
Call Center Settings and Enhancements ......................................................................................... 287
Queue Statistics ............................................................................................................................... 289
Agent Details ............................................................................................................................. 290
Login Record ............................................................................................................................. 291
Pause Log ................................................................................................................................. 291
Global Queue Settings ..................................................................................................................... 292
Switchboard ...................................................................................................................................... 294

PICKUP GROUPS .................................................................................. 297

Configure Pickup Groups ................................................................................................................. 297
Configure Pickup Feature Code ....................................................................................................... 297

MUSIC ON HOLD ................................................................................... 299

FAX SERVER ......................................................................................... 302

Configure Fax/T.38 ........................................................................................................................... 302
Receiving Fax ................................................................................................................................... 304
Example Configuration to Receive Fax from PSTN Line .......................................................... 304
Example Configuration for Fax-To-Email .................................................................................. 306
FAX Sending ..................................................................................................................................... 308

BUSY CAMP-ON .................................................................................... 309

PRESENCE ............................................................................................ 310

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FOLLOW ME .......................................................................................... 313

SPEED DIAL .......................................................................................... 315

DISA ....................................................................................................... 316

EMERGENCY......................................................................................... 318

CALLBACK ............................................................................................ 322

BLF AND EVENT LIST........................................................................... 323

BLF ................................................................................................................................................... 323
Event List .......................................................................................................................................... 323

DIAL BY NAME ...................................................................................... 326

Dial by Name Configuration.............................................................................................................. 326
Username Prompt Customization .................................................................................................... 329
Upload Username Prompt File from Web GUI ......................................................................... 329
Record Username via Voicemail Menu ..................................................................................... 330

ACTIVE CALLS AND MONITOR ........................................................... 331

Active Calls Status ............................................................................................................................ 331
Hang Up Active Calls ........................................................................................................................ 333
Call Monitor ...................................................................................................................................... 333

CALL FEATURE CODES ....................................................................... 335

Feature Codes .................................................................................................................................. 335
Parking Lot ....................................................................................................................................... 341
Call Park ........................................................................................................................................... 342
Park a Call................................................................................................................................. 342
Retrieve Parked Call ................................................................................................................. 343
Monitor Call Park CID Name Information (GXP21xx Phones Only) ......................................... 343
Call Recording .................................................................................................................................. 343
Enable Spy ....................................................................................................................................... 344
Shared Call Appearance (SCA) ........................................................................................................ 344
Announcement ................................................................................................................................. 348

PBX SETTINGS ..................................................................................... 351

PBX Settings/General Settings ........................................................................................................ 351
PBX Settings/RTP Settings .............................................................................................................. 353

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 RTP Settings ............................................................................................................................. 353
Payload ..................................................................................................................................... 354
PBX Settings/Voice Prompt Customization ...................................................................................... 355
Record New Custom Prompt .................................................................................................... 355
Upload Custom Prompt ............................................................................................................. 355
Download All Custom Prompt ................................................................................................... 356
Call Failure Tone Settings ................................................................................................................. 357
SIP Trunk Prompt Tone ............................................................................................................. 357
General Call Failure Tone ......................................................................................................... 358
PBX Settings/Jitter Buffer ................................................................................................................. 358
PBX Settings/Recordings Storage ................................................................................................... 359
PBX Settings/NAS ............................................................................................................................ 361

SIP SETTINGS ....................................................................................... 362

SIP Settings/General ........................................................................................................................ 362
SIP Settings/MISC ............................................................................................................................ 362
SIP Settings/Session Timer .............................................................................................................. 363
SIP Settings/TCP and TLS ............................................................................................................... 364
SIP Settings/NAT .............................................................................................................................. 365
SIP Settings/TOS.............................................................................................................................. 365
Transparent Call-Info header ............................................................................................................ 366

IAX SETTINGS ....................................................................................... 368

IAX Settings/General ........................................................................................................................ 368
IAX Settings/Registration .................................................................................................................. 368
IAX Settings/Security ........................................................................................................................ 369

INTERFACE SETTINGS......................................................................... 370

GDMS SETTINGS .................................................................................. 373

API CONFIGURATION ........................................................................... 376

HTTPS API (New) ............................................................................................................................. 376
HTTPS API (Old) .............................................................................................................................. 377
CDR Real-time Output Settings ....................................................................................................... 378

CTI SERVER .......................................................................................... 380

ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) ............ 381

CRM INTEGRATION .............................................................................. 382

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 SugarCRM ........................................................................................................................................ 382
vTigerCRM ........................................................................................................................................ 383
ZohoCRM ......................................................................................................................................... 385
Salesforce CRM ............................................................................................................................... 387
ACT! CRM ........................................................................................................................................ 388

PMS INTEGRATION............................................................................... 390

HMobile PMS Connector .................................................................................................................. 390
HSC PMS ......................................................................................................................................... 391
Mitel PMS ......................................................................................................................................... 392
PMS API ........................................................................................................................................... 393
Connecting to PMS ........................................................................................................................... 394
PMS Features ................................................................................................................................... 395
Room Status ............................................................................................................................. 395
Wake Up Service ...................................................................................................................... 396
Mini Bar ..................................................................................................................................... 397

WAKEUP SERVICE ............................................................................... 400

Wakeup Service using Admin Login ................................................................................................. 400
Wakeup Service from User Portal .................................................................................................... 401
Wakeup Service using Feature Code ............................................................................................... 402

ANNOUNCEMENTS CENTER ............................................................... 403

Announcements Center Settings ...................................................................................................... 403
Group Settings .................................................................................................................................. 404

QUEUEMETRICS INTEGRATION .......................................................... 406

API Configuration Parameters .......................................................................................................... 406

STATUS AND REPORTING ................................................................... 408

PBX Status ....................................................................................................................................... 408
Trunks ....................................................................................................................................... 409
Extensions................................................................................................................................. 410
Interfaces Status ....................................................................................................................... 411
System Status .................................................................................................................................. 412
General ..................................................................................................................................... 412
Network ..................................................................................................................................... 412
Space Usage............................................................................................................................. 413
Resource Usage ....................................................................................................................... 414
System Events .................................................................................................................................. 415
Alert Events List ........................................................................................................................ 415

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 Alert Log .................................................................................................................................... 417
Alert Contact ............................................................................................................................. 419
CDR .................................................................................................................................................. 420
CDR Improvement .................................................................................................................... 425
Downloaded CDR File .............................................................................................................. 425
CDR Export Customization ....................................................................................................... 426
Statistics .................................................................................................................................... 427
Recording Files ......................................................................................................................... 429
API Configuration ...................................................................................................................... 429

USER PORTAL ...................................................................................... 431

Basic Information .............................................................................................................................. 433
Personal Data ................................................................................................................................... 433
Value-added Features ...................................................................................................................... 433
Fax Sending .............................................................................................................................. 433
Call Queue ................................................................................................................................ 433
Wakeup Service ........................................................................................................................ 433
CRM User Settings ................................................................................................................... 433

MAINTENANCE ..................................................................................... 434

User Management ............................................................................................................................ 434
User Information ....................................................................................................................... 434
Custom Privilege ....................................................................................................................... 435
Concurrent Multi-User Login ..................................................................................................... 438
Login Settings ................................................................................................................................... 439
Change Password ..................................................................................................................... 439
Login Security ........................................................................................................................... 440
Operation Log ................................................................................................................................... 441
Upgrading ......................................................................................................................................... 443
Upgrading Via Network ............................................................................................................. 444
Upgrading Via Local Upload ..................................................................................................... 445
No Local Firmware Servers ...................................................................................................... 447
Backup .............................................................................................................................................. 447
Backup/Restore ........................................................................................................................ 448
Data Sync.................................................................................................................................. 450
Restore Configuration from Backup File ................................................................................... 452
System Cleanup/Reset ..................................................................................................................... 452
Reset and Reboot ..................................................................................................................... 452
Cleaner ...................................................................................................................................... 453
USB Disk/SD Card File Management ....................................................................................... 457
System Recovery ............................................................................................................................. 458

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 Syslog ............................................................................................................................................... 460
Network Troubleshooting .................................................................................................................. 461
Ethernet Capture ....................................................................................................................... 461
IP Ping ....................................................................................................................................... 462
Traceroute ................................................................................................................................. 462
Signaling Troubleshooting ................................................................................................................ 463
Analog Record Trace ................................................................................................................ 463
Service Check .................................................................................................................................. 466
Network Status ................................................................................................................................. 466

EXPERIENCING THE UCM6200 SERIES IP PBX ................................. 468

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 Table of Tables

Table 1: Technical Specifications ........................................................................................................................ 43

Table 2: UCM6200 Equipment Packaging .......................................................................................................... 46
Table 3: LCD Menu Options ................................................................................................................................ 51
Table 4: UCM6202/UCM6204 LED Indicators .................................................................................................... 52
Table 5: UCM6208 LED Indicators ...................................................................................................................... 52
Table 6: HTTP Server Settings ............................................................................................................................ 58
Table 7: UCM6200 Network Settings→Basic Settings........................................................................................ 59
Table 8: UCM6200 Network Settings→802.1X ................................................................................................... 67
Table 9: UCM6200 Network Settings→Static Routes ......................................................................................... 68
Table 10: UCM6200 Network Settings→Port Forwarding ................................................................................... 70
Table 11: UCM6200 System Settings→Network Settings→OpenVPN® ............................................................ 72
Table 12: UCM6200 Firewall→Static Defense→Current Service ....................................................................... 76
Table 13: Typical Firewall Settings ...................................................................................................................... 76
Table 14: Firewall Rule Settings .......................................................................................................................... 77
Table 15: UCM6200 Firewall Dynamic Defense ................................................................................................. 78
Table 16: Fail2Ban Settings ................................................................................................................................ 80
Table 17: Time Auto Updating ............................................................................................................................. 91
Table 18: Create New Office Time ...................................................................................................................... 93
Table 19: Create New Holiday............................................................................................................................. 95
Table 20: Email Settings ...................................................................................................................................... 96
Table 21: Email Log ........................................................................................................................................... 101
Table 22: Auto Provision Settings ..................................................................................................................... 107
Table 23: Global Policy Parameters – Localization ........................................................................................... 113
Table 24: Global Policy Parameters – Phone Settings ..................................................................................... 113
Table 25: Global Policy Parameters – Contact List ........................................................................................... 114
Table 26: Global Policy Parameters – Maintenance ......................................................................................... 116
Table 27: Global Policy Parameters – Network Settings .................................................................................. 118
Table 28: Global Policy Parameters – Customization ....................................................................................... 118
Table 29: Global Policy Parameters – Communication Settings....................................................................... 119
Table 30: Create New Template ........................................................................................................................ 121
Table 31: Create New Model Template ............................................................................................................. 122
Table 32: SIP Extension Configuration Parameters – Basic Settings ............................................................... 139
Table 33: SIP Extension Configuration Parameters – Media ............................................................................ 141
Table 34: SIP Extension Configuration Parameters – Features ....................................................................... 142
Table 35: SIP Extension Configuration Parameters – Specific Time ................................................................ 146
Table 36: IAX Extension Configuration Parameters→Basic Settings ............................................................... 147
Table 37: IAX Extension Configuration Parameters→Media ............................................................................ 148
Table 38: IAX Extension Configuration Parameters→Features ........................................................................ 148

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Table 39: IAX Extension Configuration Parameters→Specific Time................................................................. 150
Table 40: FXS Extension Configuration Parameters→Basic Settings .............................................................. 150
Table 41: FXS Extension Configuration Parameters→Media ........................................................................... 152
Table 42: FXS Extension Configuration Parameters→Features ....................................................................... 153
Table 43: FXS Extension Configuration Parameters→Specific Time ............................................................... 155
Table 44: Batch Add SIP Extension Parameters ............................................................................................... 155
Table 45: Batch Add IAX Extension Parameters ............................................................................................... 158
Table 46: SIP extensions Imported File Example ............................................................................................. 163
Table 47: IAX extensions Imported File Example ............................................................................................. 165
Table 48: FXS Extensions Imported File Example ............................................................................................ 167
Table 49: Analog Trunk Configuration Parameters ........................................................................................... 176
Table 50: PSTN Detection for Analog Trunk ..................................................................................................... 182
Table 51: Create New SIP Trunk ....................................................................................................................... 183
Table 52: SIP Register Trunk Configuration Parameters .................................................................................. 185
Table 53: SIP Peer Trunk Configuration Parameters ........................................................................................ 188
Table 54: Create New IAX Trunk ....................................................................................................................... 191
Table 55: IAX Register Trunk Configuration Parameters .................................................................................. 191
Table 56: IAX Peer Trunk Configuration Parameters ........................................................................................ 193
Table 57: SLA Station Configuration Parameters ............................................................................................. 198
Table 58: Outbound Route Configuration Parameters ...................................................................................... 201
Table 59: Outbound Routes/PIN Group ............................................................................................................ 206
Table 60: Inbound Rule Configuration Parameters ........................................................................................... 211
Table 61: Conference Room Configuration Parameters ................................................................................... 223
Table 62: Conference Settings .......................................................................................................................... 225
Table 63: Conference Caller IVR Menu ............................................................................................................ 228
Table 64: Conference Schedule Parameters .................................................................................................... 231
Table 65: Contact Group Parameters ............................................................................................................... 236
Table 66: Video Conference room Configuration Parameters .......................................................................... 239
Table 67: Video Conference Schedule Parameters .......................................................................................... 241
Table 68: IVR Configuration Parameters .......................................................................................................... 249
Table 69: Voicemail Settings ............................................................................................................................. 261
Table 70: Voicemail IVR Menu .......................................................................................................................... 262
Table 71: Voicemail Email Settings ................................................................................................................... 264
Table 72: Voicemail Group Settings .................................................................................................................. 265
Table 73: Ring Group Parameters .................................................................................................................... 266
Table 74: Restrict Calls Configuration Parameters ........................................................................................... 273
Table 75: Multicast Paging Configuration Parameters ...................................................................................... 275
Table 76: 2-way Intercom Configuration Parameters ........................................................................................ 276
Table 77: 1-way Paging Configuration Parameters .......................................................................................... 277
Table 78: Announcement Paging Configuration Parameters ............................................................................ 278
Table 79: Private Intercom Configuration Parameters ...................................................................................... 279

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Table 80: Schedule Paging / Intercom Settings ................................................................................................ 281
Table 81: Call Queue Configuration Parameters .............................................................................................. 282
Table 82: Static Agent Limitation ....................................................................................................................... 286
Table 83: Call Center Parameters ..................................................................................................................... 288
Table 84: Global Queue Settings ...................................................................................................................... 292
Table 85: Switchboard Parameters ................................................................................................................... 295
Table 86: FAX/T.38 Settings .............................................................................................................................. 302
Table 87: SIP Presence Status ......................................................................................................................... 311
Table 88: Follow Me Settings ............................................................................................................................ 314
Table 89: Follow Me Options ............................................................................................................................. 314
Table 90: DISA Settings .................................................................................................................................... 317
Table 91: Emergency Numbers Parameters ..................................................................................................... 320
Table 92: Callback Configuration Parameters................................................................................................... 322
Table 93: Event List Settings ............................................................................................................................. 324
Table 94: UCM6200 Feature Codes ................................................................................................................. 335
Table 95 : Parking Lot ....................................................................................................................................... 341
Table 96: Add SCA Private Number .................................................................................................................. 347
Table 97: Editing the SCA Number ................................................................................................................... 348
Table 98: Announcement Parameters ............................................................................................................... 349
Table 99: Internal Options/General ................................................................................................................... 351
Table 100: Internal Options/RTP Settings ......................................................................................................... 353
Table 101: Internal Options/Payload ................................................................................................................. 354
Table 102: Internal Options/Jitter Buffer ............................................................................................................ 358
Table 103: NAS Settings ................................................................................................................................... 361
Table 104: SIP Settings/General ....................................................................................................................... 362
Table 105: SIP Settings/Misc ............................................................................................................................ 362
Table 106: SIP Settings/Session Timer ............................................................................................................. 363
Table 107: SIP Settings/TCP and TLS .............................................................................................................. 364
Table 108: SIP Settings/NAT ............................................................................................................................. 365
Table 109: SIP Settings/ToS .............................................................................................................................. 365
Table 110: IAX Settings/General ....................................................................................................................... 368
Table 111: IAX Settings/Registration ................................................................................................................. 368
Table 112: IAX Settings/Static Defense ............................................................................................................ 369
Table 113: PBX Interface Settings .................................................................................................................... 370
Table 114: API Configuration Parameters ......................................................................................................... 376
Table 115: New API Supported Queries ............................................................................................................ 376
Table 116: API Configuration Parameters (Old) ................................................................................................ 377
Table 117: CDR Real-time Output Settings ....................................................................................................... 378
Table 118: SugarCRM Settings ......................................................................................................................... 382
Table 119: vTigerCRM Settings ......................................................................................................................... 384
Table 120: ZohoCRM Settings .......................................................................................................................... 386

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Table 121: Salesforce Settings.......................................................................................................................... 387
Table 122: PMS Supported Features ................................................................................................................ 390
Table 123: PMS Basic Settings ......................................................................................................................... 394
Table 124: PMS Wake up Service ..................................................................................................................... 396
Table 125: Create New Mini Bar ....................................................................................................................... 397
Table 126: Create New Maid ............................................................................................................................. 398
Table 127: Wakeup Service .............................................................................................................................. 401
Table 128: Max Wakeup Members .................................................................................................................... 401
Table 129: Announcements Center Settings ..................................................................................................... 403
Table 130: Group Settings................................................................................................................................. 404
Table 131: QueueMetrics Configuration Parameters ........................................................................................ 407
Table 132: Trunk Status .................................................................................................................................... 409
Table 133: Extension Status.............................................................................................................................. 410
Table 134: Interface Status Indicators ............................................................................................................... 411
Table 135: System Status→General ................................................................................................................. 412
Table 136: System Status→Network................................................................................................................. 413
Table 137: Alert Events ..................................................................................................................................... 415
Table 138: Alert Contact .................................................................................................................................... 419
Table 139: CDR Filter Criteria ........................................................................................................................... 420
Table 140: Automatic Download Settings .......................................................................................................... 424
Table 141: CDR Statistics Filter Criteria ............................................................................................................ 428
Table 142: API Configuration Files .................................................................................................................... 430
Table 143: User Management→Create New User ............................................................................................ 435
Table 144: Operation Log Column Header ....................................................................................................... 442
Table 145: Network Upgrade Configuration ...................................................................................................... 444
Table 146: Data Sync Configuration ................................................................................................................. 451
Table 147: Cleaner Configuration ..................................................................................................................... 455
Table 148: USB/SD Card Files Cleanup ........................................................................................................... 458
Table 149: Ethernet Capture ............................................................................................................................. 461

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 Table of Figures

Figure 1: UCM6202 Front View ........................................................................................................................... 46

Figure 2: UCM6202 Back View ........................................................................................................................... 47
Figure 3: UCM6204 Front View ........................................................................................................................... 47
Figure 4: UCM6204 Back View ........................................................................................................................... 48
Figure 5: UCM6208 Front View ........................................................................................................................... 49
Figure 6: UCM6208 Back View ........................................................................................................................... 49
Figure 7: UCM6202 Web GUI Login Page .......................................................................................................... 53
Figure 8: UCM6200 Setup Wizard ...................................................................................................................... 54
Figure 9: UCM6200 Web GUI Language ............................................................................................................ 56
Figure 10: Web GUI Search Bar ......................................................................................................................... 56
Figure 11: UCM6202 Network Interface Method: Route ..................................................................................... 63
Figure 12: UCM6202 Network Interface Method: Switch .................................................................................... 64
Figure 13: UCM6202 Network Interface Method: Dual ....................................................................................... 64
Figure 14: DHCP Client List ................................................................................................................................ 65
Figure 15: Add MAC Address Bind ..................................................................................................................... 65
Figure 16: Batch Add MAC Address Bind ........................................................................................................... 66
Figure 17: UCM6200 Using 802.1X as Client ..................................................................................................... 66
Figure 18: UCM6200 Using 802.1X EAP-MD5 ................................................................................................... 67
Figure 19: UCM6204 Static Route Sample ......................................................................................................... 69
Figure 20: UCM6204 Static Route Configuration................................................................................................ 70
Figure 21: Create New Port Forwarding ............................................................................................................. 71
Figure 22: UCM6200 Port Forwarding Configuration ......................................................................................... 71
Figure 23: GXP2160 Web Access using UCM6202 Port Forwarding ................................................................. 72
Figure 24: Open VPN® feature on the UCM6200 .............................................................................................. 73
Figure 25: Register Domain Name on noip.com ................................................................................................. 74
Figure 26: UCM6200 DDNS Setting ................................................................................................................... 75
Figure 27: Using Domain Name to Connect to UCM6200 .................................................................................. 75
Figure 28: Create New Firewall Rule .................................................................................................................. 77
Figure 29: Configure Dynamic Defense .............................................................................................................. 79
Figure 30: Fail2ban Settings ............................................................................................................................... 80
Figure 31: TLS Security ...................................................................................................................................... 82
Figure 32: SSH Access ....................................................................................................................................... 82
Figure 33: LDAP Server Configurations .............................................................................................................. 83
Figure 34: Default LDAP Phonebook DN ............................................................................................................ 84
Figure 35: Default LDAP Phonebook Attributes .................................................................................................. 84
Figure 36: LDAP Server→LDAP Phonebook ...................................................................................................... 85
Figure 37: Add LDAP Phonebook ....................................................................................................................... 85
Figure 38: Edit LDAP Phonebook ....................................................................................................................... 85

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Figure 39: Import Phonebook.............................................................................................................................. 86
Figure 40: Phonebook CSV File Format ............................................................................................................. 86
Figure 41: LDAP Phonebook After Import ........................................................................................................... 87
Figure 42: Export Selected LDAP Phonebook .................................................................................................... 87
Figure 43: LDAP Client Configurations ............................................................................................................... 88
Figure 44: GXP2170 LDAP Phonebook Configuration ....................................................................................... 90
Figure 45: Set Time Manually ............................................................................................................................. 92
Figure 46: Create New Office Time ..................................................................................................................... 93
Figure 47: Settings→Time Settings→Office Time ............................................................................................... 94
Figure 48: Create New Holiday ........................................................................................................................... 95
Figure 49: Settings→Time Settings→Holiday ..................................................................................................... 96
Figure 50: UCM6200 Email Settings ................................................................................................................... 98
Figure 51: Email Templates ................................................................................................................................. 99
Figure 52: Conference Schedule Template ....................................................................................................... 100
Figure 53: Email Send log ................................................................................................................................. 101
Figure 54: Email Logs ....................................................................................................................................... 102
Figure 55: PBX Settings→Recordings Storage ................................................................................................ 102
Figure 56: Recordings Storage Prompt Information ......................................................................................... 103
Figure 57: Recording Storage Category ........................................................................................................... 103
Figure 58: Zero Config Configuration Architecture for End Point Device ......................................................... 105
Figure 59: UCM6200 Zero Config ..................................................................................................................... 106
Figure 60: Auto Provision Settings .................................................................................................................... 107
Figure 61: Auto Discover ................................................................................................................................... 109
Figure 62: Auto Discover other subnets ............................................................................................................ 110
Figure 63: Discovered Devices ......................................................................................................................... 110
Figure 64: Device list - CSV file sample ............................................................................................................ 110
Figure 65: Managing Discovered Devices .........................................................................................................111
Figure 66: Global Policy Categories ................................................................................................................. 113
Figure 67: Edit Global Template ........................................................................................................................ 121
Figure 68: Edit Model Template ........................................................................................................................ 124
Figure 69: OEM Models .................................................................................................................................... 125
Figure 70: Template Management .................................................................................................................... 126
Figure 71: Upload Model Template Manually .................................................................................................... 126
Figure 72: Create New Device .......................................................................................................................... 127
Figure 73: Manage Devices .............................................................................................................................. 128
Figure 74: Edit Device ....................................................................................................................................... 128
Figure 75: Edit Customize Device Settings ....................................................................................................... 130
Figure 76: Modify Selected Devices - Same Model .......................................................................................... 131
Figure 77: Modify Selected Devices - Different Models .................................................................................... 132
Figure 78: Device List in Zero Config ................................................................................................................ 133
Figure 79: Zero Config Sample - Global Policy................................................................................................. 134

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Figure 80: Zero Config Sample - Device Preview 1 .......................................................................................... 135
Figure 81: Zero Config Sample - Device Preview 2 .......................................................................................... 136
Figure 82: Zero Config Sample - Device Preview 3 .......................................................................................... 137
Figure 83: Create New Device .......................................................................................................................... 138
Figure 84: Manage Extensions ......................................................................................................................... 160
Figure 85: Export Extensions ............................................................................................................................ 162
Figure 86: Import Extensions ............................................................................................................................ 162
Figure 87: Import File ........................................................................................................................................ 163
Figure 88: Import Error ...................................................................................................................................... 169
Figure 89: Extension Details ............................................................................................................................. 170
Figure 90: E-mail Notification - Prompt Information .......................................................................................... 170
Figure 91: Account Registration Information and QR Code .............................................................................. 171
Figure 92: LDAP Client Information and QR Code ........................................................................................... 171
Figure 93: Multiple Registrations per Extension ............................................................................................... 172
Figure 94: Extension - Concurrent Registration ................................................................................................ 172
Figure 95: SMS Message Support .................................................................................................................... 173
Figure 96: Edit Extension Group ....................................................................................................................... 174
Figure 97: Select Extension Group in Outbound Route .................................................................................... 175
Figure 98: UCM6200 FXO Tone Settings ......................................................................................................... 180
Figure 99: UCM6200 PSTN Detection .............................................................................................................. 180
Figure 100: UCM6200 PSTN Detection: Auto Detect ....................................................................................... 181
Figure 101: UCM6200 PSTN Detection: Semi-Auto Detect ............................................................................. 181
Figure 102: Trunk Group ................................................................................................................................... 194
Figure 103: Trunk Group Configuration ............................................................................................................ 195
Figure 104: DOD extension selection ............................................................................................................... 196
Figure 105: Edit DOD ........................................................................................................................................ 196
Figure 106: Fax Sending DOD .......................................................................................................................... 197
Figure 107: DOD Import .................................................................................................................................... 197
Figure 108: SLA Station .................................................................................................................................... 198
Figure 109: Enable SLA Mode for Analog Trunk ............................................................................................... 199
Figure 110: Analog Trunk with SLA Mode Enabled ........................................................................................... 199
Figure 111: SLA Example - SLA Station ............................................................................................................ 200
Figure 112: SLA Example - MPK Configuration ................................................................................................ 200
Figure 113: Country Codes ............................................................................................................................... 205
Figure 114: Blacklist Import/Export ................................................................................................................... 206
Figure 115: Create New PIN Group .................................................................................................................. 207
Figure 116: PIN Members ................................................................................................................................. 207
Figure 117: Outbound PIN ................................................................................................................................ 208
Figure 118: CDR Record ................................................................................................................................... 208
Figure 119: Importing PIN Groups from CSV files ............................................................................................ 209
Figure 120: Incorrect CSV File .......................................................................................................................... 209

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Figure 121: CSV File Format ............................................................................................................................ 210
Figure 122: CSV File Successful Upload .......................................................................................................... 210
Figure 123: Inbound Route feature: Prepend ................................................................................................... 216
Figure 124: Inbound Route - Multiple Mode...................................................................................................... 217
Figure 125: Inbound Route - Multiple Mode Feature Codes............................................................................. 218
Figure 126: Inbound Route - Route-Level Mode .............................................................................................. 218
Figure 127: Global Inbound Mode .................................................................................................................... 219
Figure 128: Inbound Mode - Default Mode ....................................................................................................... 220
Figure 129: Inbound Mode - Mode 1 ................................................................................................................. 220
Figure 130: Import/Export Inbound Route ......................................................................................................... 220
Figure 131: Blacklist Configuration Parameters................................................................................................ 222
Figure 132: Blacklist csv File............................................................................................................................. 222
Figure 133: Conference .................................................................................................................................... 226
Figure 134: Conference Invitation from Web GUI ............................................................................................. 227
Figure 135: Google Service Settings→OAuth2.0 Authentication ...................................................................... 229
Figure 136: Google Service→New Project ....................................................................................................... 230
Figure 137: Google Service→Create New Credential ...................................................................................... 230
Figure 138: Google Service→OAuth2.0 Login.................................................................................................. 231
Figure 139: Conference Schedule .................................................................................................................... 234
Figure 140: Contact Group Parameters ............................................................................................................ 235
Figure 141: Conference Recording ................................................................................................................... 237
Figure 142: Conference Call Statistics .............................................................................................................. 237
Figure 143: Conference Report on Web ........................................................................................................... 237
Figure 144: Conference Report on CSV ........................................................................................................... 238
Figure 145: Video Conference Basic settings ................................................................................................... 240
Figure 146: Video Conference Schedule .......................................................................................................... 242
Figure 147: Enabling WebRTC Feature ............................................................................................................ 243
Figure 148: Enabling WebRTC on Extensions ................................................................................................. 243
Figure 149: Grandstream Wave Interface ......................................................................................................... 244
Figure 150: IPVT SIP Trunk page ..................................................................................................................... 245
Figure 151 - Peer Trunk to IPVT ....................................................................................................................... 245
Figure 152 - IPVT Mode .................................................................................................................................... 246
Figure 153 - IPVT Outbound Pattern ................................................................................................................ 246
Figure 154 - IPVT Outbound Strip..................................................................................................................... 247
Figure 155: Create New IVR ............................................................................................................................. 248
Figure 156: Key Pressing Events ...................................................................................................................... 251
Figure 157: Black/White List ............................................................................................................................. 252
Figure 158: Click on Prompt to Create IVR Prompt .......................................................................................... 253
Figure 159: Language Settings for Voice Prompt ............................................................................................. 254
Figure 160: Voice Prompt Package List ............................................................................................................ 255
Figure 161: New Voice Prompt Language Added ............................................................................................. 255

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Figure 162: Upload Voice Prompts Package .................................................................................................... 256
Figure 163: Record New IVR Prompt ............................................................................................................... 257
Figure 164: Upload IVR Prompt ........................................................................................................................ 258
Figure 165: Download All Custom Prompt ........................................................................................................ 258
Figure 166: Voicemail Settings.......................................................................................................................... 260
Figure 167: Voicemail Email Settings ............................................................................................................... 264
Figure 168: Voicemail Group............................................................................................................................. 265
Figure 169: Ring Group ..................................................................................................................................... 266
Figure 170: Ring Group Configuration .............................................................................................................. 268
Figure 171: Sync LDAP Server option .............................................................................................................. 270
Figure 172: Manually Sync LDAP Server ......................................................................................................... 271
Figure 173: Ring Group Remote Extension ...................................................................................................... 271
Figure 174: Restrict Calls .................................................................................................................................. 272
Figure 175: Multicast Paging............................................................................................................................. 274
Figure 176: 2-way Intercom .............................................................................................................................. 275
Figure 177: 1-way Paging ................................................................................................................................. 276
Figure 178: Announcement Paging ................................................................................................................... 277
Figure 179: Private Intercom ............................................................................................................................. 279
Figure 180: Page/Intercom Group Settings ...................................................................................................... 280
Figure 181: Schedule Paging/Intercom page.................................................................................................... 280
Figure 182: Creating a scheduled paging/intercom call .................................................................................... 281
Figure 183: Call Queue ..................................................................................................................................... 282
Figure 184: Static Agents limit........................................................................................................................... 286
Figure 185: Agent Login Settings ...................................................................................................................... 287
Figure 186: Call Queue Statistics ..................................................................................................................... 289
Figure 187 : Automatic Download Settings - Queue Statistics ......................................................................... 290
Figure 188: Agent details .................................................................................................................................. 291
Figure 189: Login Record .................................................................................................................................. 291
Figure 190: Pause Log ...................................................................................................................................... 292
Figure 191: Global Queue Settings ................................................................................................................... 292
Figure 192: Switchboard Summary ................................................................................................................... 294
Figure 193: Call Queue Switchboard ................................................................................................................ 295
Figure 194: Queue Agent .................................................................................................................................. 296
Figure 195: Edit Pickup Group .......................................................................................................................... 297
Figure 196: Edit Pickup Feature Code .............................................................................................................. 298
Figure 197: Music On Hold Default Playlist ...................................................................................................... 299
Figure 198: Play Custom Prompt ...................................................................................................................... 300
Figure 199: Information Prompt ........................................................................................................................ 300
Figure 200: Record Custom Prompt ................................................................................................................. 301
Figure 201: Fax Settings ................................................................................................................................... 302
Figure 202: Configure Analog Trunk without Fax Detection ............................................................................. 304

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Figure 203: Configure Extension for Fax Machine: FXS Extension ................................................................. 305
Figure 204: Configure Extension for Fax Machine: Analog Settings ................................................................ 305
Figure 205: Configure Inbound Rule for Fax ..................................................................................................... 306
Figure 206: Create Fax Extension .................................................................................................................... 306
Figure 207: Inbound Route to Fax Extension ................................................................................................... 307
Figure 208: List of Fax Files .............................................................................................................................. 307
Figure 209: Fax Sending in Web GUI ............................................................................................................... 308
Figure 210: Fax Send Progress ........................................................................................................................ 308
Figure 211: SIP Presence Configuration ........................................................................................................... 310
Figure 212: SIP Presence Feature Code .......................................................................................................... 311
Figure 213: Presence Status CDR .................................................................................................................... 312
Figure 214: Edit Follow Me ............................................................................................................................... 313
Figure 215: Speed Dial Destinations ................................................................................................................ 315
Figure 216: List of Speed Dial ........................................................................................................................... 315
Figure 217: Create New DISA........................................................................................................................... 316
Figure 218: Emergency Number Configuration ................................................................................................ 319
Figure 219: 911 Emergency Sample ................................................................................................................. 321
Figure 220: Create New Event List ................................................................................................................... 324
Figure 221: Create Dial by Name Group .......................................................................................................... 326
Figure 222: Configure Extension First Name and Last Name .......................................................................... 327
Figure 223: Dial By Name Group In IVR Key Pressing Events ........................................................................ 328
Figure 224: Dial By Name Group In Inbound Rule ........................................................................................... 329
Figure 225: Status→PBX Status→Active Calls - Ringing ................................................................................. 331
Figure 226: Status→PBX Status→Active Calls – Call Established .................................................................. 331
Figure 227: Call Connection less than half hour ............................................................................................... 332
Figure 228: Call Connection between half an hour and one hour .................................................................... 332
Figure 229: Call Connection more than one hour ............................................................................................. 333
Figure 230: Configure to Monitor an Active Call ............................................................................................... 333
Figure 231: Enable/Disable Feature codes ....................................................................................................... 340
Figure 232: Parking Lot ..................................................................................................................................... 341
Figure 233: New Parking Lot ............................................................................................................................ 341
Figure 234: Monitored call park CID name ....................................................................................................... 343
Figure 235: Download Recording File from CDR Page .................................................................................... 344
Figure 236: Download Recording File from Recording Files Page ................................................................... 344
Figure 237: Enabling SCA option under Extension’s Settings .......................................................................... 345
Figure 238: SCA Number Configuration ........................................................................................................... 345
Figure 239: SCA Private Number Configuration ............................................................................................... 346
Figure 240: SCA Options .................................................................................................................................. 346
Figure 241: Announcement settings ................................................................................................................. 349
Figure 242: Announcement ............................................................................................................................... 350
Figure 243: Record New Custom Prompt ......................................................................................................... 355

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Figure 244: Upload Custom Prompt ................................................................................................................. 356
Figure 245: Download All Custom Prompt ........................................................................................................ 356
Figure 246: SIP Trunk Prompt Tone .................................................................................................................. 357
Figure 247: General call Failure Prompts ......................................................................................................... 358
Figure 248: Settings→Recordings Storage ...................................................................................................... 359
Figure 249: Recordings Storage Prompt Information ....................................................................................... 360
Figure 250: Recording Storage Category ......................................................................................................... 360
Figure 251: Transparent Call-Info ..................................................................................................................... 367
Figure 252: FXS Ports Signaling Preference .................................................................................................... 370
Figure 253: FXO Ports ACIM Settings .............................................................................................................. 370
Figure 254: DAHDI Settings .............................................................................................................................. 372
Figure 255: GDMS Developer Mode Button ..................................................................................................... 373
Figure 256: GDMS API Credentials .................................................................................................................. 373
Figure 257: GDMS Settings .............................................................................................................................. 374
Figure 258: UCM on GDMS .............................................................................................................................. 375
Figure 259: UCM SIP Extensions on GDMS .................................................................................................... 375
Figure 260: CTI Server Listening port ............................................................................................................... 380
Figure 261: SugarCRM Basic Settings ............................................................................................................. 382
Figure 262: CRM User Settings ........................................................................................................................ 383
Figure 263: vTigerCRM Basic Settings ............................................................................................................. 384
Figure 264: CRM User Settings ........................................................................................................................ 385
Figure 265: ZohoCRM Basic Settings ............................................................................................................... 385
Figure 266: CRM User Settings ........................................................................................................................ 386
Figure 267: Salesforce Basic Settings .............................................................................................................. 387
Figure 268: Salesforce User Settings ............................................................................................................... 388
Figure 269: Enabling ACT! CRM ....................................................................................................................... 388
Figure 270: Enabling CRM on the User Portal.................................................................................................. 389
Figure 271: UCM & PMS interaction ................................................................................................................. 391
Figure 272: UCM & HSC PMS interaction ........................................................................................................ 392
Figure 273: UCM & Mitel PMS interaction ........................................................................................................ 392
Figure 274: Enable PMSAPI ............................................................................................................................. 393
Figure 275: Create New Room ......................................................................................................................... 395
Figure 276: Room Status .................................................................................................................................. 395
Figure 277: Add batch rooms ............................................................................................................................ 396
Figure 278: Create New Wake Up Service ....................................................................................................... 396
Figure 279: Wakeup Call executed ................................................................................................................... 397
Figure 280: Create New Mini Bar ...................................................................................................................... 397
Figure 281: Create New Maid ........................................................................................................................... 398
Figure 282: Create New Consumer Goods ....................................................................................................... 398
Figure 283: Mini Bar .......................................................................................................................................... 399
Figure 284: Create New Wakeup Service ......................................................................................................... 400

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Figure 285: Wakeup Service Feature Code ...................................................................................................... 402
Figure 286: Announcements Center ................................................................................................................. 403
Figure 287: Announcements Center Group Configuration ................................................................................ 404
Figure 288: Announcements Center Code Configuration ................................................................................. 405
Figure 289: Announcements Center Example .................................................................................................. 405
Figure 290: QueueMetrics configuration ........................................................................................................... 406
Figure 291: Status→PBX Status ....................................................................................................................... 408
Figure 292: Trunk Status ................................................................................................................................... 409
Figure 293: Extension Status ............................................................................................................................ 410
Figure 294: UCM6204 Interfaces Status ........................................................................................................... 411
Figure 295: System Status→Storage Usage .................................................................................................... 414
Figure 296: System Status→Resource Usage ................................................................................................. 414
Figure 297: System Events→Alert Events Lists: Disk Usage ........................................................................... 416
Figure 298: System Events→Alert Events Lists: External Disk Usage............................................................. 416
Figure 299: System Events→Alert Events Lists: Memory Usage ..................................................................... 417
Figure 300: System Events→Alert Events Lists: System Crash ....................................................................... 417
Figure 301: System Events→Alert Log ............................................................................................................. 418
Figure 302: Filter for Alert Log .......................................................................................................................... 418
Figure 303: CDR Filter ...................................................................................................................................... 420
Figure 304: Call Report ..................................................................................................................................... 422
Figure 305: Call Report Entry with Audio Recording File.................................................................................. 424
Figure 306: Automatic Download Settings ........................................................................................................ 424
Figure 307: CDR Report ................................................................................................................................... 425
Figure 308: Detailed CDR Information .............................................................................................................. 425
Figure 309: Downloaded CDR File Sample ...................................................................................................... 425
Figure 310: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ........................................ 426
Figure 311: Downloaded CDR File Sample - Source Channel and Dest Channel 2 ........................................ 426
Figure 312: CDR Export File data ..................................................................................................................... 427
Figure 313: CDR Statistics ................................................................................................................................ 428
Figure 314: CDR→Recording Files................................................................................................................... 429
Figure 315: Edit User Information by Super Admin .......................................................................................... 431
Figure 316: User Portal Login ........................................................................................................................... 432
Figure 317: User Portal Layou .......................................................................................................................... 432
Figure 318: User Management Page Display ................................................................................................... 434
Figure 319: Create New User ........................................................................................................................... 434
Figure 320: User Management – New Users.................................................................................................... 435
Figure 321: Assign Backup permission to "Admin" users ................................................................................. 436
Figure 322: General User.................................................................................................................................. 437
Figure 323: Create New Custom Privilege ........................................................................................................ 438
Figure 324: Multiple User Operation Error Prompt ........................................................................................... 438
Figure 325: Change Password.......................................................................................................................... 439

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Figure 326: Login Timeout Settings .................................................................................................................. 441
Figure 327: Operation Logs .............................................................................................................................. 442
Figure 328: Operation Logs Filter ..................................................................................................................... 443
Figure 329: Network Upgrade ........................................................................................................................... 444
Figure 330: Local Upgrade................................................................................................................................ 445
Figure 331: Upgrading Firmware Files .............................................................................................................. 446
Figure 332: Reboot UCM6200 .......................................................................................................................... 446
Figure 333: Create New Backup ....................................................................................................................... 448
Figure 334: Backup / Restore ........................................................................................................................... 449
Figure 335: Local Backup ................................................................................................................................. 450
Figure 336: Data Sync ...................................................................................................................................... 451
Figure 337: Restore UCM6200 from Backup File ............................................................................................. 452
Figure 338: Reset and Reboot .......................................................................................................................... 453
Figure 339: Cleaner .......................................................................................................................................... 454
Figure 340: USB/SD Card Files Cleanup .......................................................................................................... 458
Figure 341: UCM6202 Recovery Web Page ..................................................................................................... 459
Figure 342: Recovery Mode .............................................................................................................................. 459
Figure 343: Syslog Server................................................................................................................................. 460
Figure 344: Ethernet Capture............................................................................................................................ 461
Figure 345: Ping ................................................................................................................................................ 462
Figure 346: Traceroute ...................................................................................................................................... 463
Figure 347: Troubleshooting Analog Trunks ..................................................................................................... 464
Figure 348: A Key Dial-up FXO ......................................................................................................................... 465
Figure 349: SVIP Record Trace ........................................................................................................................ 465
Figure 350: Service Check ................................................................................................................................ 466
Figure 351: Network Status............................................................................................................................... 467

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 DOCUMENT PURPOSE

The intent of this document is to provide device administrators an overview of the specifications and features
of the Grandstream UCM6200 IPPBX system. To learn more about the UCM6200, please visit
http://www.grandstream.com/support to download additional guides.

This guide covers following main topics:

• Product overview • Speed Dial

• Installation • DISA
• Getting started • Emergency
• System settings • Callback
• Provisioning • BLF and event list
• Extensions • Dial by name
• Extension groups • Active calls and monitor
• Analog Trunks • Call features
• VoIP Trunks • Call recording
• SLA station • CTI Server
• Call routes • Asterisk manager interface (AMI)
• Conference • CRM integration
• Video Conference • PMS integration
• IVR • GDMS
• Voice prompt • API
• Voicemail • QueueMetrics Integration
• Ring group • Wakeup service
• Paging and intercom group • Announcements center
• Call queue • Status and reporting
• Pickup groups • CDR (Call Details Record)
• Music On Hold • User Portal
• Fax Server • Upgrading and maintenance
• Busy camp-on • Backup/restore
• Presence • Troubleshooting
• Follow me

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 CHANGE LOG

This section documents significant changes from previous versions of the UCM6200 user manuals. Only major
new features or major document updates are listed here. Minor updates for corrections or editing are not
documented here.

Firmware Version 1.0.20.38

• Added support for FQDN and IP address format in the Realm for Digest Authentication field. [Realm For
Digest Authentication]
• Added Enable Record Feature Code option to queue settings. [Enable Record Feature Code]

Firmware Version 1.0.20.34

• Added the ability to restrict calls between extensions. [RESTRICT CALLS]

• Added the option to select the transport method information displayed in extension information emails in the
System Settings→Email Settings→Email Template→Edit Extension Template page. [Email Templates]
• Added configuration options User Information in Fax Header and Fax Header Information to the Call
Features Fax/T.38Fax Settings page. This will give users the option to send a special header in SIP fax
messages. [User Information in Fax Header][Fax Header Information]
• Added command to delete call recordings after downloading them. [recapi]
• Added DTMF Start Threshold and DTMF End Threshold options to the PBX Settings Interface Settings
DAHDI Settings page. [DAHDI Settings]
• Added External Disk Status alert event for monitoring external storage connection status. [System Events]
• The language of column titles in exported CDR reports and statistics reports will now be based on the UCM’s
display language. [Downloaded CDR File]
• Added 3 more room statuses and corresponding prompts: Room Cleared, Room Not Cleaned, and Room
Closed. [Room Status]
• Updated Hmobile Mini Bar request. [HMobile PMS Connector]
• Added the Ringback Tone option for inbound calls. [Special ringing tone]
• The value configured in the Server User Agent Value field will now replace the whole User-Agent header
value instead of just the “UCM6xxx” part. [Server User Agent]
• Added the ability to import/export a CSV file of the configured DODs for a specified trunk. [DOD
Import/Export]

Firmware Version 1.0.20.31

• Added parameters x-gscall-type and x-gs-group-name to the ACK From header sent by the UCM for calls
made to ring groups and queues to improve compatibility with non-Grandstream endpoints.
• Added GMT-4:00 (Atlantic Standard Time) and GMT+1:00 (Casablanca) to Time Settings [Time Zone]
• To improve UCM flash storage lifespan, syslog will no longer be written to UCM internal storage if a syslog
server is configured. [Syslog]

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Firmware Version 1.0.20.28

• Added HTTPS API commands to list, add, remove, and update analog trunks, SLA trunks, and digital trunks.
[HTTPS API (New)]
• Added HTTP/HTTPS protocol option for sending real-time CDR output. [Delivery Method]
• Hidden the option “Allow Guest Calls”.

Firmware Version 1.0.20.23

• No major changes.

Firmware Version 1.0.20.22

• No major changes.

Firmware Version 1.0.20.20

• Increased the maximum limit of IVR entries to 500 and inbound routes to 5000. [IVR] [Inbound Routes]
• Added support for full compliance with Kari’s law and Ray Baum’s act. [EMERGENCY]
• Increased the maximum number of outbound routes to 500. [Outbound Routes]
• Increased the maximum limit of SIP trunks to 200. [VoIP trunks]

Firmware Version 1.0.20.17

Major Enhancements

• [Announcement]
o Added Announcement feature. [Announcement]

• [Backup/Restore]
o Added NAS as a backup/restore location. [Backup/Restore]

• [Conference]
o Added conference contact groups. [Contact Group]
o Added conference call statistics and reports. [Conference Call Statistics]

• [CRM]
o Added support for Zoho CRM v2 API. [ZohoCRM]

• [GDMS]
o Added GDMS SIP account syncing. [GDMS SETTINGS]

• [HTTPS API]
o Add a new HTTPS API. [HTTPS API (New)]

• [Maintenance]

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 o Added the ability to download files on external storage from the USB Disk/SD Card File Management
page. [USB Disk/SD Card File Management]
o Added several more conditions for cleaning files and CDR. [Cleaner]

• [Queue]
o Added the ability to customize the keys used for virtual queue. [Virtual Queue Callback Key Settings]
o Improved queue statistics page. [Queue Statistics]
o Added a Welcome Prompt option to the Edit Queue page. [Welcome Prompt]
o Added QueueMetrics support. [QUEUEMETRICS INTEGRATION]

• [Paging]
o Added Private Intercom paging type (GSC3510 only). [Configure Private Intercom]

• [PMS]
o Added the ability to backup voicemail recordings upon guest check-out. [Connecting to PMS]

• [Call Routing]
o Added Outbound Route CID option to the Extensions/Trunk→Outbound Routes page. [Outbound
Route CID]

• [VoIP Trunks]
o Users can now dial into IPVT meeting rooms via peer trunks. [IPVIDEOTALK MEETINGS]
o Added support for DOD to be assigned to UCM's Fax Sending feature. [Direct Outward Dialing
(DOD)]

• [Zero Config]
o Auto Discover can now search for devices located on subnets added to the Subnet Whitelist.
[Discovery]

Other Enhancements

• [Active Calls]
o Updated Parked calls which will now be displayed in the Active Calls page. [Active Calls Status]

• [AMI]
o AMI username can no longer exceed 64 characters. [ASTERISK MANAGER INTERFACE
(RESTRICTED ACCESS)]

• [Announcement Center]
o Announcement Center group name and numbers now have a character limit of 64. [Group Settings]

• [Basic Calls]
o Added International Call Prefix. [PBX SETTINGS]
o After an attended transfer is completed, endpoints will now be updated with the CID of the other
party instead of keeping the transferor’s CID.
o Added Block collect calls option in SIP settings→General settings. [Block Collect Calls]

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• [Backup/Restore]
o Added event logs and email notifications for scheduled SFTP backup and data sync results. [Alert
Events List]

• [Call Completion]
o Updated the number of CC agents to be limited by the extensions allowed number of concurrent
registrations. [CC Mode] [BUSY CAMP-ON]

• [CDR]
o Added Extension Groups as a CDR filter. [Table 139: CDR Filter Criteria]

• [Conference]
o Added a field to configure the Kick Prompt Interval. [Kick Warning Interval]
o Added support for Conference participant names to be announced when joining/leaving the
conference even when the conference is on hold. [Announce Callers]
o Extended the Meeting duration to be scheduled up to 8 hours. [Meeting Duration]
o Conference extension number can no longer be used as the conference password if Strong
Password is enabled. [Password]

• [CRM]
o Added “Add Unknown Number” option for ACT! CRM. [ACT! CRM]
o Added https://www.zohoapis.eu to the ZohoCRM Server Address dropdown list. [ZohoCRM]

• [Dial by Name]
o Added the ability to upload a custom Dial by Name prompt. [Custom Prompt]

• [DOD]
o Added DOD names and numbers character limit to 32. [Direct Outward Dialing (DOD)]

• [Email Settings]
o Added new variable ${CONFR_MEMBERS}, which shows conference participant details. [Email
Templates]
o Added the following email type filters to the Email Send Log page: Send Fax, Call Queue Statistics,
Conference Report. [Email Send Log]

• [Emergency Calls]
o Users can now strip the same amount of numbers as the emergency number length itself. [Strip]

• [Eventlist]
o Added a search bar for eventlists. [Event List]

• [Extensions]
o The Extension and Auth ID fields now support plus signs (+). [Extension] [Auth ID]
o Enable CC option added as batch edit extension option. [Enable CC]
o Subnet mask can now be specified when configuring subnets that can register to the extension.
[ACL Policy] [Local Network Address]

• [GS Wave Web]

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 o A conference participant's CID number will now be displayed instead of extension number if
configured. [Wave WebRTC Video Calling & Conferencing]
o Improved extension searching when transferring. [Wave WebRTC Video Calling & Conferencing]
o An error message will now appear when the maximum number of allowed registrations has been
reached. [Wave WebRTC Video Calling & Conferencing]
o Hosts can now generate a link that can be used to invite others to the conference. [Wave WebRTC
Video Calling & Conferencing]
• [IVR]
o Added support to upload more than one welcome prompt for IVR. [Welcome Prompt]

• [LDAP]
o Plus signs (+) is now supported when creating contacts and receiving calls with “+” in the CID. [LDAP
Phonebook]
o LDAP Client CA Cert field no longer supports .ca file uploads. The following file types are supported:
.crt .der .pem. [LDAP Client CA cert]
o The Username field will no longer require "cn=" to precede username. [Username]
o Character limit changed for the following fields:
• LDAP Server→Root Password: 32 [LDAP Server Configurations]
• LDAP Server→Root DN: 64 [LDAP Server Configurations]
• Phonebook Download Configurations→Username: 64 [Username]
o LDAP now supports downloading phonebooks from domains. [Server Address]
o Added space support to the LDAP Client Username. [Username]

• [Maintenance]
o Added Conference Call Statistics Report Cleaner. [Conference Call Statistics Report Cleaner]

• [OpenVPN®]
o Added option Allow Weak SSL Ciphers. [Allow Weak SSL Ciphers]

• [Paging]
o Added an option to allow scheduled paging to play during holidays. [Include Holidays]
o “=” is no longer supported in the Alert-Info Header field. [Paging/Intercom Group Settings]

• [Parking]
o Ring All Callback on Timeout will not be available if Forward to Destination on Timeout is enabled.
[Ring All Callback on Timeout]
o The Failover Destination field now has a character limit of 32. [Failover Destination]

• [PMS]
o The Username and Guest Category Code columns in the Room Status page can now be sorted.
[Room Status]

• [Queue]
o Agents can now view recordings of their calls in their user portal. [USER PORTAL]
o A queue can now have 3 chairmen to manage it. [Queue Chairman]
o Callers can no longer use feature codes in established callbacks. [Enable Feature Codes]

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• [Ring Group]
o The character limit of the email address fields for ring group voicemail boxes has been changed
from 256 to 128. [Enable Destination]
o Added the option to skip busy members when receiving an incoming call. [Skip Busy Agent]

• [SIP Settings]
o The TLS Self-signed CA field no longer supports .ca file uploads. The following file types are
supported: .ca .crt .der .pem. [TLS Self-Signed CA]

• [Recording]
o MoH will now be recorded when a party is on hold. [Call Recording]

• [Routing]
o Users can now add patterns to the blacklist to restrict calls from several numbers. [Outbound
Blacklist]
o Inbound route imports/exports now have the following columns: Fax Detection, Fax Intelligent
Routing, and Fax Destination. [Inbound Route: Import/Export Inbound Route]
o Users can now export PIN Groups. [PIN Groups]
o The PIN Groups with Privilege Level option has been added to allow PIN groups to work alongside
Privilege Level and Filter on Source Caller ID settings. [PIN Groups with Privilege Level]

• [System Information]
o Added duplex type and network port connection speed information to System Information→Network
page. [Speed] [Duplex Mode]

• [Time Settings]
o Users can now configure holidays for the next 4 years. [Year]
o Added time condition Office Time and Out of Holiday. [Time Condition]
o Default NTP server is now pool.ntp.org. [Remote NTP Server]

• [User Management]
o Added the following custom privileges [Custom Privilege]:
• CDR Records
• CDR Statistics
• CDR Recordings
• Voice Prompt
• Inbound Routes
• Outbound Routes
o Added the option to change Super Administrator username in the Change Information page.
[Change Username]
o Regular users will now be logged out automatically if an admin resets their extension. [USER
PORTAL]

• [Voicemail]
o Voicemail group member limit increased from 16 to 27. [Member]

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 o The voicemail system will now mention if a voicemail is from a ring group. [Enable Destination]

• [Voice Prompt]
o Voice prompt package upload file size limit increased to 50MB. [Upload Language Package]
o Custom voice prompt file name character limits for the Voice Prompt and Announcement Center
pages have been changed to 100. [Upload Custom Prompt] [ANNOUNCEMENTS CENTER]
[Announcement]

• [VoIP Trunks]
o PAI headers now have a character limit of 64 and can only contain alphanumeric characters and/or
special characters #*-_+. [PAI Header]
o When editing a register trunk or trunk group and TLS is selected in the Transport field, users can
now select between SIP and SIPS URI scheme. [SIP URI Scheme When Using TLS]
o Domain names can now be sent in the request URL by configuring the From Domain field. [From
Domain]

• [Security]
o Added support to specify minimum/maximum TLS version. [TLS Security]

Firmware Version 1.0.19.29

• Updated extension DND response. [General Call Failure Tone]

• Added feature codes for remote management of extension call forwarding. [Remote Call Forward
Enable]
• The wakeup service prompt now has an option to set a wakeup for the next day. [Wakeup Service]

Firmware Version 1.0.19.27

• Added new option Email-to-Fax Subject Format to allow email subjects to consist of only the fax number.
[Email-to-Fax Subject]
• Added new options Auto Record to automatically record emergency calls and Send Recording File to
send these recordings to a configured email address. [Auto Record]
• Increased concurrent registration limit to 300 for UCM62xx’s wave web interface. [Wave WebRTC Video
Calling & Conferencing]
• Added Emergency Recordings page to display recordings of emergency calls. [EMERGENCY]
• Added a toggle checkbox to enable/disable Announcement Paging. [Enable]
• Queue position will now be announced to the caller upon entering the queue. [Enable Position
Announcement]
• Added new option Enable Hold time Announcement to announce estimated wait times to caller. [Enable
Hold time Announcement]
• Added a new option “Block the Backward Collect Call” to automatically block collect calls/reverse charge
calls. [Block ]

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 • Added Chile time zone. [Automatic Date and Time]
• Added IPVT Mode option to VoIP Peer Trunk [IPVT Mode].

Firmware Version 1.0.19.21

• Optimized system performance for UCM6202 and UCM6204 models, which now support up to 50 and
75 concurrent calls, respectively. [Maximum Call Capacity]
• Added ACT! CRM support. [ACT! CRM]
• Added Email-to-Fax Blacklist/Whitelist. [Email-to-Fax Blacklist/Whitelist]
• Added digit prepending and stripping support for Emergency calls. [EMERGENCY]
• Added new functionalities to Grandstream Wave Web. [Wave WebRTC Video Calling & Conferencing]
• Added LDAPS protocol support. [Technical Specifications] [LDAP Server]
• Added support for .conf and .ovpn files for OpenVPN. [OpenVPN®]
• Added Announcement Paging type. [Configure Announcement Paging]
• Added PMS API as a PMS options. [PMS API]
• Added ability to clear agents call counters. [Reset Agent Call Counter - Enable]
• Added Set CallerID option. [Set Caller ID Info]
• Added ability to monitor and change the inbound mode of individual inbound routes and toggle/monitor
them via BLF. [Inbound Mode] [Inbound Route: Route-Level Mode] [Inbound Route: Inbound Mode BLF
Monitoring]
• Added search functionality to the web portal to quickly find settings. [Web GUI Search Bar]
• Added HT818 model template to Zero Config Models Templates. [PROVISIONING]
• Added the ability to associate differently named OEM models with their original GS models for
provisioning purposes. [OEM device models]

Firmware Version 1.0.18.13

• No major changes.

Firmware Version 1.0.18.12

• CDR API configuration page moved from CDR to Value-Added Features.

• Added ability to upload voice prompt files via API.
• Added ability to transfer to custom numbers, not just extensions using Grandstream Wave Web [Wave
WebRTC Video Calling & Conferencing]

Firmware Version 1.0.18.9

• Added new AMI commands. [ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)]

• Added ability to customize the data columns included in exported CDR reports. [CDR Export
Customization]
• Added ability to view a specified extension’s membership in Call Queues/Ring Groups and other details.
[Extension Details]

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 • Added ability to disable audio files for Music on Hold. [MUSIC ON HOLD]
• Added Timeout Destination and Ring-All Callback on Timeout options to Parking Lot page. [Parking Lot]
• Added inbound rule importing/exporting functionality. [Inbound Route: Import/Export Inbound Route]
• Added GS Wave WebRTC video calling and video conferencing functionality. [Wave WebRTC Video
Calling & Conferencing]

Firmware Version 1.0.17.16

• Added RTX codec support. [Technical Specifications] [Codec Preference]

• Emergency calls will no longer be logged into CRM servers. [CRM INTEGRATION]
• Added batch extension resetting functionality. [Batch Extension Resetting Functionality]
• Added FAX Resend Support. [Enable Fax Resend] [Max Resend Attempts]
• Fail2ban now supports up to 20 whitelist entries. [Fail2Ban Whitelist]
• Added FXO Frequency Tolerance option to Analog Hardware page. [FXO Frequency Tolerance]
• Added NAS support for call recording backup. [NAS]
• Added paging/intercom scheduling functionality. [Configure a Scheduled Paging/Intercom]
• Added Multicast Paging support.
• Added Failover Destination under call parking options. [Failover Destination]
• Added Timeout Callback Ringing All options under call parking options. [Ring All Callback on Timeout]
• Added ability to view CID of parked calls on VPKs/MPKs configured as monitored call park. [Monitor Call
Park CID Name Information (GXP21xx Phones Only)]
• Added support for endpoint call forwarding under ring group. [Endpoint Call Forwarding Support]
• Added batch operations and searching functionality to the Inbound/Outbound Blacklist pages.
Operations include deleting rules and importing/exporting entire blacklists. [Outbound Blacklist][Blacklist
Configurations]
• Added ability to monitor and toggle inbound routing modes via BLF. [Inbound Mode BLF Monitoring]
• Added Shared Call Appearance functionality. [Shared Call Appearance]
• Added Forward HOLD Requests option under Misc page. [Forward HOLD Requests]
• Added support for Trunk Groups. [Trunk Groups]
• Added Forward Voicemail to Peered UCMs option. [Forward Voicemail to Peered UCMs]
• Added a new Voicemail Password field under Voicemail settings. [Voicemail Password]
• Added Catalan language support under voice prompt. [VOICE PROMPT]
• Added DOD Name field under VoIP trunk settings. [Direct Outward Dialing (DOD)]
• Changed the default value of Max Wait Time to 60. [Max Wait Time]
• Added “SCA” action type to filtering CDR options. [Action Type]
• Added a prompt asking whether to delete all recording files or not on CDR. [CDR]
• Added a 64-character limit to Conference → Extension field. [Extension]
• Added a 3-character limit to Extension Incrementation field. [Extension Incrementation]
• Added a maximum limit of 500 entries to the IVR blacklist and whitelist. [IVR]
• Added character restrictions to LDAP Number Attributes field to ensure correct input. [LDAP Client
Configurations]
• Added maximum limit of 99,999 to parking timeout. [Parking Timeout]

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 • Added a 63-character limit to the Room Number field. [Room Number]
• Editing an already executed wakeup service will automatically change the service’s status to
“Programmed”. [Action Status]
• Adding UCM disconnection when an SSH client changes the password for the connected user.
• LCD display will show “Recovery Mode” instead of “No Provision” when the UCM is in recovery mode.
[System Recovery]
• Capture files saved on external devices will have “capture” prepended to file names.
• Added dashes and + characters to Fax, Home Number, and Mobile Phone Number fields.
• Updated external number to accepts only letters, numbers and special characters. [External Number]
• Added a 64-charcter limit to Custom Presence Status field. [Custom Presence Status]
• Increased the character limit for provider name field to 64 characters during creation. [Provider Name]
• Updated From Domain field to accept special characters. [From Domain]
• Username Prompt filename character limit changed to 18. [Username Prompt Customization]
• Increased the character limit for the wakeup service name field to 64 characters. [Wakeup Service]
• Users can now upload custom prompts directly to pages without having to get redirected to the Voice
Prompt → Custom Prompt page. [Custom Prompt]
• Added character restrictions to device firmware file names. [firmware file name]
• Added Email-to-Fax support. [Email-to-Fax]

Firmware Version 1.0.16.20

• Added an option to enable/disable remote voicemail access function. [Voicemail Remote Access]

Firmware Version 1.0.16.18

• Added support for emergency calls. [EMERGENCY]

• Added support for vTigerCRM [vTigerCRM]
• Added support for ZohoCRM [ZohoCRM v2 ]
• Added support for HSC PMS [HSC PMS]
• Added support for Direct Callback. [Direct Callback] [Direct Callback] [Direct Callback]
• Added option to disable call waiting under extensions. [Enable Call Waiting]
• Added option to passthrough SIP PAI Header. [Passthrough PAI Header]
• Added option to copy device settings in zero config. [Managing discovered devices]
• Added support for call failure prompts customization. [General Call Failure Tone]
• Added option to automatically request HTTP server certificate. [Certificate Options]
• Added support to include more menus on user custom privilege. [Custom Privilege]
• Added possibility to delete CDR records based on search result. [CDR]
• Added option to send caller to specific destination when queue is empty. [CALL QUEUE]
• Added XML file type support for importing phonebooks. [LDAP Phonebook]
• Added ability to customize the LDAP name and number attributes in the Phonebook Download
Configurations page [LDAP Client Configurations]
• Added option to download Call Queue Statistics. [Queue Statistics]

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 • Added option to choose DNS mode during calls. [DNS mode]
• Added ability to batch add devices by importing a CSV file [PROVISIONING]
• Added a Copy Device Configuration option that allows users to copy the configuration of one device to
another device [PROVISIONING]
• Added ability to divide a parking lot extension into multiple lots [Parking Lot]
• Added option to reset extension settings. [Reset single extension]

Firmware Version 1.0.15.16

• 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. [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. [voice message responses]
• 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]
• Added CDR records separation [CDR separation]

Firmware Version 1.0.14.24

• Added protection to prevent HTTP rogue login.

Firmware Version 1.0.14.23

• Restored ability to view voicemail count in the Extension/Trunk overview.

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 • 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

• 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.
[PROVISIONING]
• Improved seamless transfer privilege control. [Seamless transfer privilege control]
• Added RTP Keep-alive support. [RTP Keep-alive]

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 • 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 Incrementation]
• 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. [Save to External Storage]
• 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 Disk/SD Card File Management]
• Added support for A Key Dial-up FXO. [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
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. [Space Usage]

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 • Added support foe Ring Group/Call Queue/IVR Display Option for Caller ID. [Replace Display Name |
Replace | 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]
• 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 Se]
• 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 product and void
the manufacturer warranty.
-------------------------------------------------------------------------------------------------------------------------------------- ------

This document is subject to change without notice. The latest electronic version of this user manual is available
for download here:
http://www.grandstream.com/support

Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any
purpose without the express written permission of Grandstream Networks, Inc. is not permitted.

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 PRODUCT OVERVIEW

Technical Specifications

The following table resumes all the technical specifications including the protocols / standards supported, voice
codecs, telephony features, languages and upgrade/provisioning settings for UCM6200 series.
Table 1: Technical Specifications

Interfaces
Analog Telephone FXS 2x RJ11 ports with lifeline support
Ports Each port supports 2 REN
• UCM6202: 2 ports
PSTN Line FXO Ports • UCM6204: 4 ports
• UCM6208: 8 ports
UCM6202/6204/6208:
Network Interfaces 1x Gigabit WAN port
1x Gigabit LAN port with PoE Plus (IEEE 802.3at-2009)
NAT Router Yes
Peripheral Ports USB 2.0, 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 128ms tail-length carrier-grade Line Echo Cancellation with NLP Packetized Voice
Capabilities Protocol Unit, dynamic jitter buffer, modem detection, and auto-switch to G.711.
G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC (30ms
Voice and Fax Codecs
only), GSM, AAL2-G.726-32, RTX, ADPCM; T.38
Video Codecs H.264, H.265, H.263, H.263+, VP8
QoS Layer 3 QoS, Layer 2 QoS
Signaling and Control
DTMF Methods Inband, RFC4733, and SIP INFO
TFTP/HTTP/HTTPS, auto-discovery and auto-provisioning of Grandstream IP
Provisioning Protocol and
endpoints via ZeroConfig (DHCP Option 66/multicast SIP SUBSCRIBE/mDNS),
Plug-and-Play
Eventlist between local and remote trunk
TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP, SSH,
Network Protocols
HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS, LDAP/LDAPS
Disconnect Methods Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current

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 Disconnect, Busy Tone
Security
Media SRTP, TLS1.2, HTTPS, SSH
Advanced Defense Fail2ban, alert events, whitelist, blacklist and strong password requirement.
Physical
• Output: 12VDC, 1.5A
Universal Power Supply
• Input: 100-240VAC, 50-60Hz
• UCM6202/6204: 226mm (L) x 155mm (W) x 34.5mm (H)
Dimensions
• UCM6208: 440mm (L) x 185mm (W) x 44mm (H)
• Operating: 32 - 104oF / 0 - 40oC, 10-90% (non-condensing)
Environmental
• Storage: 14 - 140oF / -10 - 60oC
• UCM6202/6204: Wall mount and Desktop
Mounting
• UCM6208: Rack mount and Desktop
• UCM6202: Unit weight 0.51kg, Package weight 0.94kg
Weight • 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
IVR/voice prompts for English, Chinese, British English, German, Spanish,
Multi-language Support
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
Up to 5 layers of IVR (Interactive Voice Response)
Attendant
• UCM6202:
Concurrent audio calls up to 50, concurrent WebRTC calls up to 25.
• UCM6204:
Maximum Call Capacity Concurrent audio calls up to 75, concurrent WebRTC calls up to 35.
• UCM6208:
Concurrent audio calls up to 100, concurrent WebRTC calls up to 50.
Or up to 66% performance if calls are SRTP encrypted
• UCM6202/6204 up to 500 registered SIP endpoints.
SIP Devices
• UCM6208 up to 800 registered SIP endpoints.

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 • UCM6202/6204: Up to 3 password-protected conference rooms allowing up
to 25 simultaneous PSTN or IP participants
Conference Rooms
• UCM6208: Up to 6 password-protected conference rooms allowing up to 32
simultaneous PSTN or IP participants
Call park, call forward, call transfer, DND, ring/hunt group, paging/intercom and
Call Features
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
Compliance
• 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 a 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 1x
Power Adaptor 1x
Ethernet Cable 1x
Quick Installation Guide 1x
GPL License 1x

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 power jacks located on the back of the UCM6202. It is highly
recommended to connect the other end of the plug to 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 solid
green and the LCD will display the IP address.
6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax)
to the FXS ports.

Note: The ground screw needs to be connected.

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 power jacks located on the back of the UCM6204. It is
highly recommended to connect the other end of the plug to 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
solid green and the LCD will display the IP address.
6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and
Fax) to the FXS ports.

7. Note: The ground screw needs to be connected.

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 power jacks located on the back of the UCM6208. It is highly
recommended to connect the other end of the plug to 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 solid green and the LCD will display the IP address.
6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and Fax)
to the FXS ports.

Note: The ground screw needs to be connected.

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 USB Port 8 x FXO Port LED Indicators Navigation Keys

SD Card Slot 2 x FXS Port LCD

Figure 5: UCM6208 Front View

LAN Port Wan Port

Wan Port

DC 12 V Reset 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 an 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

To get started with the UCM6200 setup process, use the following available interfaces: LCD display, LED
indicators, and web portal.

• The LCD display shows hardware, software, and network information and can be navigated via the DOWN
and OK buttons next to the display. From here, users can configure basic network settings, run diagnostic
tests, and factory reset.
• The LED indicators at the front of the device provides interface connection and activity status.
• The web portal (may also be referred to as web UI in this guide) is the primary method of configuring the
UCM.

This section will provide step-by-step instructions on how to use these interfaces to quickly set up the UCM and
start making and receiving calls with it.

Use the LCD Menu

• Idle Screen
Once the device has booted up completely, the LCD will show the UCM model, hardware version (e.g.,
V1.4A), and IP address. Upon menu key press timeout (15 seconds), the screen will default back to this
information. Pressing the DOWN button will show the system time.

• Menu
Pressing the OK button will show the main menu. All available menu options are found in [Table 3: LCD
Menu Options].

• Menu Navigation
Pressing the DOWN button will scroll through the menu options. Press the OK button to select an option.

• Exit
Selecting the Back option will return to the previous menu. For the Device Info, Network Info, and Web Info
screens that have no Back option, pressing the OK button will return to the previous menu.

• LCD Backlight
The LCD backlight will turn on upon button press and will go off when idle for 30 seconds.

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The following table summarizes the layout of the LCD menu of UCM.

Table 3: LCD Menu Options

• Critical Events
View Events
• Other Events

• Hardware: Hardware version number

• Software: Software version number
• P/N: Part number
Device Info
• WAN MAC: WAN side MAC address
• LAN MAC: LAN side MAC address
• Uptime: System uptime

• WAN Mode: DHCP, Static IP, or PPPoE

• WAN IP: IP address
Network Info • WAN Subnet Mask
• LAN IP: IP address
• LAN Subnet Mask

• WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE

Network Menu
• Static Route Reset: Select this to reset static route settings.

• Reboot
• Factory Reset
• LCD Test Patterns
Press DOWN and OK buttons to scroll through and select different LCD
patterns to test. Once a test is done, press the OK button to return to the
previous menu.
• Fan Mode
Select Auto or On.
Factory Menu • LED Test Patterns
All On, All Off, and Blinking are the available options. Selecting Back in the
menu will revert the LED indicators back to their actual status.
• RTC Test Patterns
Select either 2022-02-22 22:22 or 2011-01-11 11:11 to start the RTC (Real-
Time Clock) test pattern. Check the system time from either the LCD idle
screen or in the web portal System Status->System Information-
>General page. To revert back to the correct time, manually reboot the
device.

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 • Hardware Testing
Select Test SVIP to verify hardware connections within the device. The result
will display on the LCD when the test is complete.

• Protocol: Web access protocol (HTTP/ HTTPS). HTTPS is used by default.

Web Info
• Port: Web access port number, which is 8089 by default.

• Enable SSH
SSH Switch • Disable SSH
SSH access is disabled by default

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 LED Status

NETWORK
Solid: Connected
OFF: Not Connected

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 ACT
USB
Solid: Connected
SD
Flashing: Data Transferring
Phone (FXS)
OFF: Not Connected
Line (FXO)

Using the Web UI

Accessing the Web UI

The UCM's web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML pages allow users to
configure the device through a web browser such Microsoft IE (version 8+), Mozilla Firefox, Google Chrome,
etc. To access the UCM's web portal, follow the steps below:

Figure 7: UCM6202 Web GUI Login Page

1. Make sure your computer is on the same network as the UCM.

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 2. Make sure that the UCM's IP address is displayed on its LCD.

3. Enter the UCM's IP address into a web browsers' address bar. The login page should appear (please
see the above image).

4. Enter default administrator username “admin” and password.

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 logging in for the first
time. Older units have default password “admin”.

----------------------------------------------------------------------------------------------------------------------------- ----------------------

Note:
By default, the UCM6200 has Redirect From Port 80 enabled. As such, 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, and 192.168.40.167 is entered into the web browser, the web page
will be redirected to: https://192.168.40.167:8089
The option Redirect From Port 80 can be found in the System Settings→HTTP Server page.
----------------------------------------------------------------------------------------------------------------------------- ----------------------

Setup Wizard

After logging into the UCM web portal for the first time, the setup wizard will guide the user through basic
configurations such as time zone, network settings, trunks, and routing rules.

Figure 8: UCM6200 Setup Wizard

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The setup wizard can be closed and reopened at any time. At the end of the wizard, a summary of the pending
configuration changes can be reviewed before applying them.

Main Settings

There are 8 main sections in the web portal to manage various features of the UCM.

• System Status: Displays the dashboard, system information, current active calls, and network status.

• Extensions/Trunks: Manages extensions, trunks, and routing rules.

• Call Features: Manages various features of the UCM such as the IVR and voicemail.

• PBX Settings: Manages the settings related to PBX functionality such as SIP settings and interface
settings.

• System Settings: Manages the settings related to the UCM system itself such as network and security
settings.

• CDR: Contains the call detail records, statistics, and audio recordings of calls processed by the UCM.

• Value-Added Features: Manages the settings of features unrelated to core PBX functionality such as Zero
Config provisioning and CRM/PMS integrations.

• Maintenance: Manages settings and logs related to system management and maintenance such as
user management, activity logs, backup settings, upgrade settings and troubleshooting tools.

Web GUI Languages

Currently the UCM6200 series Web GUI supports English, Simplified Chinese, Traditional Chinese,
Spanish, French, Portuguese, Russian, Italian, Polish, German, Turkish and Czech.

Users can select the UCM's web UI display language in the top-right corner of the page.

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 Figure 9: UCM6200 Web GUI Language

Web GUI Search Bar

Users can search for options in the web portal with the search bar on the top right of the page.

Figure 10: Web GUI Search Bar

Saving and Applying Changes

After making changes to a page, click on the "Save" button to save them and then the "Apply Changes" button
that appears to finalize the changes. If a modification requires a reboot, a prompt will appear asking to reboot
the device.

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Setting Up an Extension

Power on the UCM6200 and your SIP endpoint. Connect both devices to the same network and follow the steps
below to set up an extension.
1. Log into the UCM web portal and navigate to Extension/Trunk->Extensions
2. Click on the "Add" button to start creating a new extension. The Extension and SIP/IAX Password
information will be used to register to this extension. To set up voicemail, the Voicemail Password will
be required.
3. To register an endpoint to this extension, go into your endpoint's web UI and edit the desired account.
Enter the newly created extension's number, SIP user ID, and password into their corresponding fields
on the endpoint. Enter the UCM's IP address into the SIP server field. If setting up voicemail, enter *97
into the Voice Mail Access Number field. This field may be named differently on other devices.
4. To access the extension's voicemail, use the newly registered extension to dial *97 and access the
personal voicemail system. Once prompted, enter the voicemail password. If successful, you will now
be prompted with various voicemail options.
5. You have now set up an extension on an endpoint.

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 SYSTEM SETTINGS

This section will explain the available system-wide parameters and configuration options on the UCM62xx. This
includes settings for the following items: HTTP server, network, OpenVPN, DDNS, LDAP server and email
server.

HTTP Server

The UCM6200's embedded web server responds to HTTP/HTTPS GET/POST requests and allows users to
configure the UCM via web browsers such as Microsoft IE, Mozilla Firefox, and Google Chrome. By default,
users can access the UCM by just typing its IP address into a browser address bar. The browser will automatically
be redirected to HTTPS using port 8089. For example, typing in "192.168.40.50" into the address bar will redirect
the browser to "https://192.168.40.50:8089". This behavior can be changed in the System Settings->HTTP
Server page.

Table 6: HTTP Server Settings

Toggles automatic redirection to UCM's web portal from port 80. If

disabled, users will need to manually add the UCM's configured
Redirect From Port 80
HTTP/HTTPS port to the server address when accessing the UCM web
portal via browser. Default is "Enabled".
Select either HTTP or HTTPS as the protocol to access the UCM's HTTP
Protocol Type server. This will also determine what is used when endpoints download
config files from the UCM via Zero Config. Default is "HTTPS".
Specifies the port number used to access the UCM HTTP server. Default
Port
is "8089".
If enabled, only the server addresses in whitelist will be able to access the
UCM's web portal. It is highly recommended to add the IP address
Enable IP Whitelist
currently used to access the UCM web page before enabling this option.
Default is "Disabled".
List of addresses that can access the UCM web portal.
Permitted IP(s)
Ex: 192.168.6.233 / 255.255.255.255
Selects the method of acquiring SSL certificates for the UCM web server.
Two methods are currently available:
Certificate Options - Upload Certificate: Upload the appropriate files from one's own PC.
- Request Certificate: Enter the domain for which to request a certificate
for from Let's Encrypt.

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 Uploads the private key for the HTTP server.

TLS Private Key

Note: Key file must be under 2MB in file size and in *.pem format. File
name will automatically be changed to "private.pem".
Uploads the certificate for the HTTP server.

TLS Cert Note: Certificate must be under 2MB in file size and in *.pem format. This
will be used for TLS connections and contains private key for the client
and signed certificate for the server.
Enter the domain to request the certificate for and click on
Domain
to request the certificate.

If the protocol or port has been changed, the user will be logged out and redirected to the new URL.

Network Settings

After successfully connecting the 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 7: 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 will be used for uplink connection. LAN port will function similarly to a regular
Method
router port.
• Switch
WAN port will be used for uplink connection. LAN port will be used as a bridge for
connections.

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 • Dual
Both WAN and LAN ports will be used for uplink connections labeled as LAN2 and
LAN1, respectively. The port selected as the Default Interface will need to have a
gateway IP address configured if it is using a static IP.

MTU Specifies the maximum transmission unit value. Default is 1500.

IPv4 Address

Preferred DNS
If configured, this will be used as the Primary DNS server.
Server

WAN (when "Method" is set to "Route")

IP Method Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

IP Address Enter the IP address for static IP settings. The default setting is 192.168.0.160.

Subnet Mask Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.

Gateway IP Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.

DNS Server 1 Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.

DNS Server 2 Enter the DNS server 2 address for static IP settings.

Username Enter the username to connect via PPPoE.

Password Enter the password to connect via PPPoE.

Layer 2 QoS
802.1Q/VLAN 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 or disable DHCP server capability. The default setting is "Yes".
Enable

DNS Server 1 Enter DNS server address 1. The default setting is 8.8.8.8.

DNS Server 2 Enter DNS server address 2. The default setting is 208.67.222.222.

Allow IP
Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.
Address From

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Allow IP
Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.
Address To

Default IP
Enter the IP lease time (in seconds). The default setting is 43200.
Lease Time

LAN (when Method is set to "Switch")

IP Method Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

IP Address Enter the IP address for static IP settings. The default setting is 192.168.0.160.

Subnet Mask Enter the subnet mask address for static IP settings. The default setting is 255.255.0.0.

Gateway IP Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.

DNS Server 1 Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.

DNS Server 2 Enter the DNS server 2 address for static IP settings.

Username Enter the username 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 If "Dual" is selected as the Method, users must select either LAN 1 (WAN port) or LAN 2
Interface (LAN port) to be used as the default interface. Default setting is LAN 2.

IP Method Select DHCP, Static IP, or PPPoE. The default setting is DHCP.

Enter the IP address for static IP settings.

IP Address
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.

Enter the gateway IP address for static IP settings when the port is assigned as default
Gateway IP
interface. The default setting is 0.0.0.0.

Enter the DNS server 1 address for static IP settings.

DNS Server 1
The default setting is 0.0.0.0.

DNS Server 2 Enter the DNS server 2 address for static IP settings.

Username Enter the username to connect via PPPoE.

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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 Enter the DNS server 2 address for static settings.

LAN (when Method is set to "Route")

Select Disable, Auto or DHCPv6.

Disable: Disables the DHCPv6 server.
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.

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 IP Prefixlen Enter the Prefix length for static settings. Default is 64

DNS Server 1 Enter the DNS server 1 address for static settings.

DNS Server 2 Enter the DNS server 2 address for static settings.

LAN 1 / LAN 2 (when Method is set to "Dual")

Default Users must select either LAN 1 (WAN port) or LAN 2 (LAN port) to be used as the default
Interface interface. Default setting 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 the sample diagram below.

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

• 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

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DHCP Client List

This page lists all the detected devices on the LAN and the IP addresses that were provided to them. Additionally,
users can manually link a MAC address to an IP address.

When devices receive IP addresses from the UCM's DHCP server, they will be listed in the System Settings-
>Network Settings->DHCP Client List page as shown below.

Figure 14: DHCP Client List

To manually add and bind a MAC address to an IP address, click on . The following menu
will then be displayed.

Figure 15: Add MAC Address Bind

Enter the device's MAC address and the IP address to bind it to. This IP address must be in the UCM's DHCP
range.

To bind multiple existing MAC addresses that are in the list to their respective IP addresses, check the boxes

next to each MAC address and click on the button. A confirmation message will

appear on the screen. Click to bind the addresses.

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 Figure 16: Batch Add MAC Address Bind

The Bind Status for the selected MAC addresses should now be changed from "Unbound" to "Bound".

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 use 802.1X
mode “EAP-MD5” on WAN port as client in the network to access Internet.

Figure 17: UCM6200 Using 802.1X as Client

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 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.

Table 8: UCM6200 Network Settings→802.1X

Select 802.1X mode. The default setting is "Disable". The supported 802.1X
modes 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
Select 802.1X client certificate from local PC and then upload.
Certificate

Static Routes

The UCM6200 provides users static routing capability that allows the device to use manually configured
routes, instead of dynamically assigned routes or the default gateway. Static routes can be used to define a

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route when no others are available or can serve as complementary routes alongside existing routes as failover
routes, 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 to edit the static route.

• Select to delete the static route.

Table 9: 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
default value is 255.255.255.255.
Netmask
Example:
255.255.255.0

Configure the IPv4 or IPv6 gateway address so that the UCM6200 can reach the
destination via this gateway. Gateway address is optional.
Gateway
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.

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 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

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.

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 Figure 20: UCM6204 Static Route Configuration

Port Forwarding

The UCM network interface supports router functionality that allows users to configure port forwarding. If the
UCM is set to "Route" in System Settings->Network Settings->Basic Settings->Method, the Port Forwarding
tab will be available.

Table 10: UCM6200 Network Settings→Port Forwarding

Specify the WAN port number or a range of WAN ports. Unlimited number of ports
can be configured.

WAN Port Note:

When set to a range, both the WAN port and LAN port must be configured with the
same range (e.g., WAN port: 1000-1005, LAN port: 1000-1005). Access from the
WAN port will be forwarded to the LAN port with the same port number.

LAN IP Specify the LAN IP address.

Specify the LAN port number or a range of LAN ports.

LAN Port
Please see the note for WAN Port.

Select protocol type "UDP Only", "TCP Only" or "TCP/UDP" for the forwarding in
Protocol Type
the selected port. The default setting is "UDP Only".

Here is an example of an environment and how to configure port forwarding to an endpoint's web portal.

• Configure the UCM as a router by selecting "Route" in System Settings->Network Settings->Basic

Settings->Method.
➢ WAN port is connected to an uplink switch with a public IP address configured (e.g., 1.1.1.1).

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 ➢ LAN port provides a DHCP pool for devices on the LAN network (gateway address is
192.168.2.1 by default).
• Connect a GXP2160 to the UCM's LAN network. It should obtain an IP address from UCM's DHCP pool.
• While still on the Network Settings page, navigate to the Port Forwarding tab.
• Click on button to start setting up port forwarding.

Figure 21: Create New Port Forwarding

WAN Port: Enter the port that will be opened on the WAN side to allow access.
LAN IP: Enter the GXP2160's IP address.
LAN Port: Enter the port that will be opened on the GXP2160 to allow access.
Protocol Type: Select the protocol to use for this port forwarding.

Figure 22: UCM6200 Port Forwarding Configuration

This will allow users to access the GXP2160 web portal from an external network.

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 Figure 23: GXP2160 Web Access using UCM6202 Port Forwarding

OpenVPN®

The UCM can be configured as an OpenVPN client.

Table 11: UCM6200 System Settings→Network Settings→OpenVPN®

Enable Enable / Disable the OpenVPN® feature.

Select OpenVPN® configuration method.

Configuration Manual Configuration: Allows to configure OpenVPN settings manually.
Method Upload Configuration File: Allows to upload. ovpn and .conf files to the UCM
and to automatically configure OpenVPN settings.

OpenVPN®
Configures the hostname/IP and port of the server. For example: 192.168.1.2:22
Server Address

OpenVPN® Specify the protocol user, user should use the same settings as used on the
Server Protocol server

Use the same setting as used on the server.

OpenVPN®
Dev TUN: Create a routed IP tunnel.
Device mode
Dev TAP: Create an Ethernet tunnel.

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 OpenVPN® Use Compress tunnel packets using the LZO algorithm on the VPN link. Do not
Compression enable this unless it is also enabled in the server config file.

Allow Weak SSL

Enable/Disable allowing Weak SSL Ciphers.
Ciphers

OpenVPN®
Specify the cryptographic cipher. Users should make sure to use the same
Encryption
setting that they are using on the OpenVPN server.
Algorithm

OpenVPN® CA Upload as SSL/TLS root certificate. This file will be renamed as ‘ca.crt’
Cert automatically.

OpenVPN®
Upload a client certificate. This file will be renamed as ‘cliend.crt’ automatically.
Client Cert

OpenVPN®
Upload a client private key. This file will be renamed as ‘client.key’ automatically.
Client Key

Figure 24: Open VPN® feature on the UCM6200

DDNS Settings

Configuring UCM DDNS settings will allow the UCM to be accessed via domain name instead of IP address.
UCM supports the following DDNS services:

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• dydns.org
• noip.com
• freedns.afraid.org
• zoneedit.com
• oray.net

Here is an example of using noip.com for DDNS.

1. The UCM must be publicly accessible in order to work with the service provider.

Figure 25: Register Domain Name on noip.com

2. Navigate to System Settings->Network Settings->DDNS Settings, check the Enable DDNS option, select
your service provider, and configure all the displayed fields.

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 Figure 26: UCM6200 DDNS Setting

3. You can now use the configured domain to access the UCM web portal.

Figure 27: Using Domain Name to Connect to UCM6200

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Security Settings

The UCM offers several methods of protection against malicious attacks and unauthorized access such as
firewall rules, connection thresholds, and Fail2ban.

To get started on configuring security settings, navigate to the System Settings->Security Settings page.

Static Defense

On the Static Defense page, users can configure firewall rules and view the ports used by various UCM
services and processes.

The following table shows a few examples of the information available on the Static Defense page.

Table 12: 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 general firewall defense mechanisms, UCM supports Ping Defense, SYN-Flood Defense, and Ping-of-
Death Defense. These can be configured separately for the WAN interface and LAN interface.

Table 13: Typical Firewall Settings

Ping Defense
If enabled, the UCM will not send ICMP messages in response to ping requests.
Enable

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 If enabled, the UCM will limit the amount of SYN packets accepted from one
source to 10 packets per second, preventing the UCM web portal from
SYN-Flood becoming inaccessible. Excess packets will be discarded.
Defense Enable
There is no need to mention the WAN and LAN parts since it is already
mentioned in sentence before the table.
Ping-of-Death
If enabled, the UCM will not be affected by ping of death attacks.
Defense Enable

In the Custom Firewall Settings section, users can create rules to accept, reject, or drop specific traffic going

through the UCM. To create a new rule, click on the button.

The following menu will appear. Here is an example firewall rule created to reject SSH access to the UCM from
the WAN interface.

Figure 28: Create New Firewall Rule

Table 14: Firewall Rule Settings

Rule Name Enter a name for the firewall rule.

Select the action for the Firewall to perform.

• ACCEPT
Action
• REJECT
• DROP

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 Select the traffic type.
• IN
Type If selected, the Interface field will appear. Users must specify the interface
that the inbound rule will be applied to.
• OUT
Select the connection type the rule will apply to.
• FTP
• SSH
• Telnet
• TFTP
Service • HTTP
• LDAP
• Custom
If selected, users must configure the Source IP and Port,Destination IP
Address and Port, and the Protocol fields that appear. If the source and
destination are left blank, the "Anywhere" values will be used.

Save the change and click on "Apply" button. Then submit the configuration by clicking on "Apply Changes" on
the upper right of the web page. The new rule will be listed at the bottom of the page with sequence number,
rule name, action, protocol, type, source, destination and operation. More operations below:

• Click on to edit the rule.

• Click on to delete the rule.

Dynamic Defense

On the Dynamic Defense page, users can configure the UCM to monitor incoming TCP connections and
prevent excessive traffic from hosts. The UCM must have "Route" configured in the System Settings-
>Network Settings->Basic Settings page. The blacklist on this page is automatically updated.

The following options are available:

Table 15: UCM6200 Firewall Dynamic Defense

Dynamic Defense
Toggle dynamic defense on and off. This is disabled by default.
Enable
Configure the blacklist update time interval (in seconds). The default setting is
Blacklist Update 120.
Interval For example, if set to "300", blocked IP addresses will not be able to establish
TCP connections with the UCM until after 300 seconds have passed.

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 Connection Configure the connection threshold. Once a host exceeds this threshold, it will
Threshold be added to the blacklist. Default setting is 100.
Allowed IPs and ports range, multiple IP addresses and port range.
Dynamic Defense For example:
Whitelist 192.168.5.100-
192.168.5.200 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.
• 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 and prevents SIP brute force attacks on the PBX system.

Once an IP address exceeds the allowed number of login or SIP authentication attempts within the configured
"Max Retry Duration" period, all SIP and HTTP requests from that IP address will be dropped, forbidding web
access and blocking further authentication attempts.

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 Figure 30: Fail2ban Settings
Table 16: 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.
Configure the duration (in seconds) for the detected host to be banned. The default
Banned Duration
setting is 600. If set to 0, the host will be always banned.

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 If a host exceeds the maximum allowed number attempts configured for Max
Max Retry Duration Retry within the configured Max Retry Duration window, the host will be
banned. The default setting is 600 seconds.
Configures the maximum number of allowed authentication failures within the
MaxRetry
configured Max Retry Duration window. The default setting is 5.
Configures the IP addresses, CIDR masks, and DNS hosts in the Fail2Ban
Fail2Ban Whitelist whitelist. Whitelisted entries will not be banned by Fail2Ban even after exceeding
the allowed number of authentication failures. Up to 20 addresses can be added.
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.
Configures the maximum number of authentication failures before the host is
MaxRetry banned. The default setting is 10. Please note that this will override the Global
Settings->MaxRetry setting.
Enables defense against excessive login attacks to the UCM’s web GUI.
Login Attack Defense
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.
Configures the maximum allowed number of failed login attempts from an IP
MaxRetry
address before it is added to the Fail2Ban blacklist.
Blacklist
Blacklist Users will be able to view the IPs that have been blocked by UCM.

TLS Security

TLS security to specify minimum and maximum TLS versions supported by the UCM.
Please log in UCM62xx web interface and go to System Settings→Security Settings→TLS Security.
By default, minimum TLS version is set to TLS1.1, and maximum TLS version is set to TLS1.2.
Supported versions are 1.0, 1.1 and 1.2

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 Figure 31: TLS Security

SSH Access

SSH access can be toggled from the UCM's webUI and physical LCD screen. The webUI option can be found
under System Settings->Security Settings->SSH Access. SSH access is disabled by default and should only be
turned on for troubleshooting and debugging.

Figure 32: SSH Access

LDAP Server

The UCM6200 has an embedded LDAP/LDAPS server for users to manage corporate phonebooks in a
centralized manner.

• The LDAP server automatically generates an initial phonebook with PBX DN "ou=pbx,dc=pbx,dc=com"
based on the UCM6200's existing extensions.
• 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

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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.

Figure 33: LDAP Server Configurations

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The UCM6200 LDAP server supports anonymous access (read-only) by default. Therefore, the LDAP client does
not have to configure username and password to access the phonebook directory. The "Root DN" and "Root
Password" (limited with 64 and 32 characters respectively) 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 34: Default LDAP Phonebook DN

Figure 35: 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.

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 Figure 36: 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.

Figure 37: 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 38: Edit LDAP Phonebook

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• Import phonebook from your computer to LDAP server

Click on “Import Phonebook” and a dialog will prompt as shown in the figure below.

Figure 39: Import Phonebook

The file to be imported must be a CSV, VCF or XML file with UTF-8 encoding. Users can open the file with
Notepad and save it with UTF-8 encoding.

Here is an example of CSV file. Please note “Account Number” and “Phonebook DN” fields are required. Users
can export a phonebook file from the UCM's LDAP phonebook section and use it as a template.

Figure 40: 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 does not 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.

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 Figure 41: LDAP Phonebook After Import

As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in LDAP
phonebook section, users cannot import contacts that have "pbx" in the Phonebook DN field of the CSV file.

• Export phonebook to your computer from UCM6200 LDAP server

Select the checkbox for the LDAP phonebook and then click on “Export Selected Phonebook” to export the
selected phonebook. The exported phonebook can be used as a record or a sample CSV, VFC or XML file for
the users to add more contacts in it and import to the UCM6200 again.

Figure 42: Export Selected LDAP Phonebook

LDAP Client Configurations

The configuration on LDAP client is useful when you use other LDAP servers. Here we provide an example on
how to configure the LDAP client on the UCM.

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Assuming the remote server base dn is “dc=pbx,dc=com”, configure the LDAP client as follows:

• LDAP Server: Enter a name for the remote LDAP server

• Server Address: Enter the IP address or domain name for remote LDAP server.
• Base DN: dc=pbx,dc=com
• Username: Enter username if authentication is required. This field cannot exceed 64 characters and
can contain space.
• Password: Enter password if authentication is required.
• Filter: Enter the filter. Ex: (|(CallerIDName=%)(AccountNumber=%))
• Port: Enter the port number. Ex:389
• LDAP Name Attributes: Enter the name attributes for remote server
• LDAP Number Attributes: Enter the number attributes for remote server

• Client type: Protocol of LDAP or LDAPS.

• LDAP Client CA cert: Upload LDAP client CA certificate, The following file types are supported: .crt .der
and .pem.

• LDAP Client Private Key: Upload LDAP client private key.

The following figure gives a sample configuration for UCM acting as a LDAP client.

Figure 43: LDAP Client Configurations

To configure Grandstream IP phones as the LDAP clients for UCM, please refer to the following example:

• Server Address: The IP address or domain name of the UCM6200

• Base DN: dc=pbx,dc=com
• Username: Please leave this field empty
• Password: Please leave this field empty
• LDAP Name Attribute: CallerIDName Email Department FirstName LastName
• LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax
• LDAP Number Filter: (AccountNumber=%)

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 • 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 33: LDAP Server Configurations].

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Figure 44: GXP2170 LDAP Phonebook Configuration

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Time settings

Automatic Date and Time

The current system time can be found under System Status->Dashboard and System Information.
To configure the UCM to automatically update time, navigate to System Settings->Time Settings->Automatic
Date and Time.
----------------------------------------------------------------------------------------------------------------------------- ----------------------

Note:
The configurations under Web GUI→Settings→Time Settings→Automatic Date and Time 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.
----------------------------------------------------------------------------------------------------------------------------- ---------------------

Table 17: Time Auto Updating

Specify the server address of the NTP server for the UCM6510 to sync
Remote NTP Server
date and time with. The default NTP server is pool.ntp.org.

If set to "Yes", the UCM's time zone can be provisioned via DHCP Option
Enable DHCP Option 2
2 from a local server automatically.

If set to "Yes", the UCM's NTP server can be provisioned via DHCP Option
Enable DHCP Option 42 42 from a local server automatically. This will override the manually
configured NTP server. The default setting is "Yes".

Select the time zone for the UCM to use.

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]

Self-Defined Time Zone

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

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 the Prime Meridian (A.K.A: International or Greenwich Meridian); If it is
negative (-), the local time zone is east.

M4.1.0,M11.1.0
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec).
The 2nd number indicates the nth iteration of the weekday: (1st Sunday,
3rd Tuesday…). Normally 1, 2, 3, 4 are used. If 5 is used, it means the last
iteration of the weekday.
The 3rd number indicates weekday: 0,1,2,..,6 ( for Sun, Mon, Tues, ...
,Sat).
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.

Set Time Manually

To manually set the time on the UCM6200, go to Web GUI→System Settings→Time Settings→Set Date and
Time. The format is YYYY-MM-DD HH:MM:SS.

Figure 45: 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 UCM can act as an NTP server for clients to sync their system times with. To configure this, navigate to
System Settings->Time Settings->NTP Server and set Enable NTP Server to "Yes". On the client side, use
the UCM's IP address or hostname as the NTP server address.

Office Time

The system administrator can define office hours which can be used as conditions for call forwarding and
inbound routing. To configure this, navigate to System Settings->Time Settings->Office Time and click on
the Add button to create office hours.

Figure 46: 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 workdays in one week.
Check this option to show advanced options. Once selected, please
Show Advanced Options
specify "Month" and "Day" below.

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 Month Select the months for office time.
Day Select the workdays 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.

Figure 47: 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

System administrators can define holidays which can be used as conditions for call forwarding and inbound
routing. To configure this, navigate to System Settings->Time Settings->Holiday and click on the Add button
to create a new holiday.

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 Figure 48: Create New Holiday

Table 19: Create New Holiday

Name Specify the holiday name to identify this holiday.

Holiday Memo Create a note for the holiday.
Select the year of the Holiday.
Year
Note: Users can now configure holidays for the next 4 years.
Month Select the month for the holiday.
Day Select the day for the holiday.
Check this option to show advanced options. If selected, please specify
Show Advanced Options
the days as holiday in one week below.
Week Select the days as holiday in one week.

Enter holiday "Name" and "Holiday Memo" for the new holiday. Then select "Month" and "Day". The system
administrator can also define days in one week as advanced options. Once done, click on "Save" and then
"Apply Change" for the holiday to take effect. The holiday will be listed in the web page as the figure shows
below.

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 Figure 49: 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
------------------------------------------------------------------------------------------------------------------------------------ --------

Email Settings

Email settings
The UCM's email module can send out alert event emails, fax (Fax-to-Email), voicemail (Voicemail-to-Email),
etc. Email settings can be configured in System 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. UCM will act as a virtual mail server, and
emails will be sent from the configured domain. There is no need to
configure SMTP server settings, and no credentials will be required.
Type However, emails sent using MTA may be considered as spam by
destination SMPT servers.
• Client: UCM will send emails to the configured SMTP server to
forward to the final destination. Credentials for the SMTP server will
be required.

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 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". This is typically the
Username
email address.
Password Password for the entered Username.
Toggles the Email-to-Fax feature.
If enabled, the UCM will monitor the configured email inbox (using
provided [Username] and [Password]) for emails with the subject
"SendFaxMail To XXX" or “XXX”.
Enable Email-to-Fax Subject example: SendFaxMail To 7200 Or 7200.
The UCM will extract the attachments of detected emails and send it to
the XXX extension by fax. The attachment must be in PDF/TIF/TIFF
format.
Note: This field will appear when using Type “Client”.
Email-to-Fax
Enables the Email to fax Blacklist/Whitelist functionality.
Blacklist/Whitelist
Specify the Email subject format for fax sending, the subject can be
Email-to-Fax Subject
either "SendFaxMail To XXX" or “XXX" with XXX the fax number.
Specify the Email address blacklist/whitelist for local extensions. This
Internal Blacklist/Whitelist feature prevents faxing from unauthorized email addresses. The
internal list includes only contacts with local extensions.
Specify the Email address blacklist/whitelist for non-local contacts. This
feature prevents faxing from unauthorized email addresses. The
External Blacklist/Whitelist external list is for non-local contacts.
Note: Multiple addresses can be separated with semicolon (;) i.e.
" XXX;YYY ".
Configure the POP/POP3 server address for the configured username
POP/POP3 Server Address
Example: pop.gmail.com
Configure the POP/POP3 server port for the configured username
POP/POP3 Server Port
Example: 995
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.

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 The following figure shows example email settings, where 192.168.6.202 is the SMTP server.

Figure 50: UCM6200 Email Settings

Once configuration is complete, click on the Save button first and then the Test button to verify that the settings
work.

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Email Templates

The UCM provides various email templates for different email notifications. Email templates can be accessed
from System Settings->Email Settings->Email Templates.

Figure 51: Email Templates

Press on to upload pictures to be used on email templates.

Press to reset all email templates to default ones.

To configure the email template, click the button under Options column, and edit the template as desired.

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 Figure 52: Conference Schedule Template

- Users have the ability to preview mail sample by clicking on .

- Click on in order to restore the default email template.

- Finally, users can click on 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.

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 Figure 53: 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

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
• Emergency calls
• Video conference Schedule
• User Password
• Alert Events
• CDR

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 • Fax
• Fax sending
• Call Queue Statistics
• Conference Reports
• PMS
• Test
• Missed calls

Email logs will be shown on bottom of the “Email Send Log” page, as shown on the following figure.

Figure 54: Email Logs

Recordings Storage

The UCM supports both automatic and manual call recordings. Recordings can be saved to UCM's local
storage, external storage (SD/USB disks), and NAS.

Figure 55: PBX Settings→Recordings Storage

• If Enable Auto Change is selected, the recording storage location will automatically change to NAS, USB
Disk, or SD card if they are available. If all storage location types are available, the priority will be NAS->USB
Disk->SD Card->Local storage.

• If “Local” is selected, the recordings will be stored in UCM62xx internal storage.

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• 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 UCM62xx.
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 56: Recordings Storage Prompt Information

Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to
be copied over.

Figure 57: Recording Storage Category

On the UCM62xx, recording files are generated and exist in 3 categories: normal call recording files,
conference recording files, and call queue recording files. Therefore, users have the following options when
select the categories to copy the files to the external device:

• Recording Files: Copy the normal recording files to the external device.
• Conference: Copy the conference recording files to the external device.
• Queue: Copy the call queue recording files to the external device.
• All: Copy all recording files to the external device.

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 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 properly manage SIP
devices.
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 58: 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 59: 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 60: Auto Provision Settings

Table 22: Auto Provision Settings

Enable or disable the zero-config feature on the PBX. The default setting
Enable Zero Config
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
UCM6200.
Enable Automatic
Configuration Assignment
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.
If enabled, when the device is discovered, the PBX will automatically
Automatically Assign
assign an extension within the range defined in "Zero Config Extension
Extension
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
range to be assigned if "Automatically Assign Extension" is enabled. The
Zero Config Extension
default range is 5000-6299. Zero Config Extension Segment range can be
Segment
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
the device's request. This feature is for the GXP series phones that support
Enable Pick Extension
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".
Specify the number of minutes to allow the phones being provisioned to
Pick Extension Period (hour)
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 immediately apply
the configuration.

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 61: 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.

Auto Discover can also search for devices located on other subnets, in condition for the subnet to be added
under Zero Config Settings> Subnet Whitelist. The method allowed to auto discover other subnets then the
UCM’s is SIP-Message like shown below.

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 Figure 62: Auto Discover other subnets

Figure 63: Discovered Devices

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 and an existing account),
otherwise the UCM will reject the file and the operation will fail:

Figure 64: Device list - CSV file sample

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Managing discovered devices

• Sorting: Press or to sort per MAC Address, IP Address, Version, Vendor, Model or Create
Config columns from lower to higher or higher to lower, respectively.

• Filter: Select a filter to display corresponding results.

▪ All: Display all discovered devices.
▪ Scan Results: Display only manually discovered devices. [Discovery]
▪ IP Address: Enter device IP and press Search button.
▪ MAC Address: Enter device MAC and press Search button.
▪ Model: Enter a model name and press Search button. Example: GXP2130.

Figure 65: Managing Discovered Devices

From the main menu of zero config, users can perform the following operations:

• Click on in order to access to the discovery menu as shown on [Discovery] section.

• Click on to add a new device to zero config database using its MAC address.

• Click on to delete selected devices from the zero-config database.

• Click on to modify selected devices.

• Click on to batch update a list of devices, the UCM on this case will send SIP
NOTIFY message to all selected devices in order to update them at once.

• Click on to reboot selected devices (the selected devices, should have been

provisioned with extensions since the phone will authenticate the server which is trying to send it reboot
command).

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 • Click on to clear all devices configurations.

• Click on to upload CSV file containing list of devices.

• Click on to copy configuration from one device to another. This can be useful for
easily replace devices and note that this feature works only between devices of same model.

All these operations will be detailed on the next sections.

Global configuration

Global policy

Global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN with the
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.
• Communication Settings: configure Email and FTP settings

Select the checkbox on the left of the parameter you would like to configure to activate the dropdown list for this
parameter.

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 Figure 66: Global Policy Categories

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
Time Format
device’s LCD.
Configure the URL or IP address of the NTP server. The SIP end device
NTP Server
may 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
Dial Plan
to user manual of the SIP devices to be provisioned for more details.
When enabled, “Do Not Disturb”, “Call Forward” and other call features can
Enable Call Features be used via the local feature code on the phone. Otherwise, the ITSP
feature code will be used.
If set to “Yes”, pressing the number key “#” will immediately dial out the
Use # as Dial Key
input digits.

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 If set to “Yes”, the phone will automatically turn on the speaker phone to
answer incoming calls after a short reminding beep, based on the SIP
Auto Answer by Call-info
Call-Info header sent from the server/proxy.
The default setting is enabled.
NAT Traversal Configure if NAT traversal mechanism is activated.
If set to “Yes”, this parameter will force random generation of both the
User Random Port
local SIP and RTP ports.
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.
• If “Manual” is selected, the LDAP configuration below will be applied to
Source 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.
Example:
Base DN
• dc=grandstream, dc=com
• ou=Boston, dc=grandstream, dc=com
Configure the bind “Username” for querying LDAP servers. The field can
Username
be left blank if the LDAP server allows anonymous binds.
Configure the bind “Password” for querying LDAP servers. The field can be
Password
left blank if the LDAP server allows anonymous binds.

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 Configure the filter used for number lookups. Please refer to user manual
Number Filter
for 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
Version
default value is 3.
Specify the “name” attributes of each record which are returned in the
LDAP search result.
Name Attribute Example:
gn
cn sn description
Specify the “number” attributes of each record which are returned in the
LDAP search result.
Number Attribute Example:
telephoneNumber
telephoneNumber Mobile
Configure the entry information to be shown on phone’s LCD. Up to 3 fields
can be displayed.
Display Name
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.
Specify the interval (in seconds) for the server to process the request and
Search Timeout
client waits for server to return. Valid range is 0 to 180. Default value is 30.
Sort Results Specify whether the searching result is sorted or not. Default setting is No.
Configure to enable LDAP number searching when receiving calls. The
Incoming Calls
default setting is No.
Configure to enable LDAP number searching when making calls. The
Outgoing Calls
default setting is No.
Configures the display name when LDAP looks up the name for incoming
Lookup Display Name call or outgoing call.
It must be a subset of the LDAP Name Attributes.
XML Phonebook
Select the source of the phonebook XML server.
• Disable
Disable phonebook XML downloading.
Phonebook XML Server
• Manual
Once selected, users need specify downloading protocol HTTP,
HTTPS or TFTP and the server path to download the phonebook XML

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 file. The server path could be IP address or URL, with up to 256
characters.
• Local UCM Server
Once selected, click on the Server Path field to upload the phonebook
XML file. Please note after uploading the phonebook XML file to the
server, the original file name will be used as the directory name and
the file will be renamed as phonebook.xml under that directory.
Phonebook Download Configure the phonebook download interval (in Minute). If set to 0,
Interval automatic download will be disabled. Valid range is 5 to 720.
Remove manually-edited If set to “Yes”, when XML phonebook is downloaded, manually added
entries on download entries will be removed.

Table 26: Global Policy Parameters – Maintenance

Upgrade and Provision

UCM can provision the firmware source to devices. The following options
are available:

Select a source to get the firmware file:

• URL
If selected, complete the configuration for the following four
parameters: “Upgrade Via”, “Server Path”, “File Prefix” and “File
Postfix”.

• Local UCM Server

Firmware can be uploaded to the UCM6200 internal storage for
Firmware Source firmware upgrade. If selected, click on “Manage Storage” icon next to
“Directory” option, upload firmware file and select directory for the end
device to retrieve the firmware file.

• Local USB Media

If selected, the USB storage device needs to be plugged into the
UCM6200 and the firmware file must be put under a folder named
“ZC_firmware” in the USB storage root directory.

• Local SD Card Media

If selected, an SD card needs to be plugged into the UCM6200 and the
firmware file must be put under a folder named “ZC_firmware” in the
USB storage root directory.
When URL is selected as firmware source, configure upgrade via TFTP,
Upgrade via
HTTP or HTTPS.

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 When URL is selected as firmware source, configure the firmware
Server Path
upgrading server path.
When URL is selected as firmware source, configure the firmware file
prefix. If configured, only the firmware with the matching encrypted prefix
File 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
File Postfix postfix. If configured, only the configuration file with the matching encrypted
postfix will be downloaded and flashed into the phone.
If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can
Allow DHCP Option 43/66
be redirected.
If enabled, the end point device will automatically upgrade if a new firmware
is detected. Users can select automatic upgrading by day, by week or by
minute.
• By week
Once selected, specify the day of the week to check HTTP/TFTP
server for firmware upgrades or configuration files changes.
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 Specifies how firmware upgrade and provision requests will be sent.
Web Access
Admin Password Configure the administrator password for admin level login.
End-User Password Configure the end-user password for the end user level login.
Web Access Mode Select HTTP or HTTPS as the web access protocol.
Web Server Port Configure the port for web access. The valid range is 1 to 65535.
Security
Enable Telnet/SSH access for the SIP end device. If the SIP end device
supports Telnet access, this option controls the Telnet access of the device;
Disable Telnet/SSH
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 Configures whether SIP messages will be included in the syslog.

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 Table 27: Global Policy Parameters – Network Settings

Basic Settings
Configures how the SIP endpoint will obtain IP addresses. The following
options are available:

• DHCP
Once selected, users can specify the Host Name (option 12) of the SIP
IP Address end device as DHCP client, and Vendor Class ID (option 60) used by
the client and server to exchange vendor class ID information.

• PPPoE
Once selected, users need specify the Account ID, Password and
Service Name for PPPoE.
Advanced Setting
Define the Layer 3 QoS parameter. This value is used for IP Precedence,
Layer 3 QoS
Diff-Serv or MPLS. Valid range is 0-63.
Layer 2 QoS Tag Assign the VLAN Tag of the Layer 2 QoS packets. Valid range is 0 -4095.
Layer 2 QoS Priority Value Assign the priority value of the Layer 2 QoS packets. Valid range is 0-7.
Configure the IP address or Domain name of the STUN server. Only non-
STUN Server
symmetric NAT routers work with STUN.
Specify how often the phone will send a blank UDP packet to the SIP server
Keep Alive Interval in order to keep the “ping hole” on the NAT router to open. Valid range is
10-160.

Table 28: Global Policy Parameters – Customization

Wallpaper
Check this option if the SIP endpoint uses a 1024x600 resolution
wallpaper.

• Source
Screen Resolution 1024 x Configure the location where wallpapers are stored.
600 • 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 endpoint uses a 800x400 resolution wallpaper.

Screen Resolution 800 x 400 • Source

Configure the location where wallpapers are stored.
• File

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 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 endpoint uses a 480x272 resolution
wallpaper.

• Source

Screen Resolution 480 x 272 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 endpoint uses a 320x240 resolution wallpaper.

• Source
Configure the location where wallpapers are stored.
Screen Resolution 320 x 240 • 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.

Table 29: Global Policy Parameters – Communication Settings

Email Settings
Check this option to configure the email settings that will be sent to the
provisioned phones:

• Server
IP address of the SMTP server

• Port

SMTP server port

• From E-Mail address

SMTP Settings
Email address

• Sender Username

Username of the sender

• Password Recovery Email

Email where recovered password will be sent

• Alarm receive Email 1

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 Email address where alarms notifications will be sent

• Alarm receive Email 1

Email address where alarms notifications will be sent

• Enable SSL

Enable SSL protocol for SMTP

FTP
Check this option to configure the FTP settings that will be sent to the
provisioned phones:

• Storage Server Type

Either FTP or Central Storage

• Server
FTP server address
FTP
• Port
FTP port to be used

• Username
FTP username

• Path
FTP Directory path

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 templates for the device. Please refer to section [Manage Devices] for more details on using the
global templates.

When creating global template, users can select the categories and the parameters under each category to be
used in the template. The global policy and the selected global template will both take effect when generating
the config file. However, the selected global template has higher priority to the global policy when it comes to the
same setting option/field. If the same option/field has different value configured in the global policy and the
selected global template, the value for this option/field in the selected global template will override the value in
global policy.

Click on "Create New Template" to add a global template. Users will see the following configurations.

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 Table 30: 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.

Figure 67: Edit Global Template

The added options will show in the list. Users can then enter or select value for each option to be used in the
global template. On the left side of each added option, users can click on to remove this option from the
template. On the right side of each option, users can click on to reset the option value to the default value.

Click on “Save” to save this global template.

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• 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 templates for the
device. Please refer to section [Manage Devices] for more details on using the model template.

For each created model template, users can assign it as default model template. If assigned as default model
template, the values in this model template will be applied to all the devices of this model. There is always only
one default model template that can be assigned at one time on the 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 31: Create New Model Template

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.

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 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 overwritten if other selected
model template has a different value for the same option.

Active Check this option to enable the model template.

• Click on to edit the model template.

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.
Note: Some devices use Alias and not P-values.

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 Figure 68: 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 including OEM
device models.

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OEM Models

Users can associate OEM device models with their original Grandstream-branded models, allowing these OEM
devices to be provisioned appropriately.

• Click on button.
• In the Source Model field, select the Grandstream device that the OEM model is based on from the dropdown
list.

• For the Destination Model and Destination Vendor field, enter the custom OEM model name and vendor
name.

• The newly added OEM model should now be selectable as an option in Model fields.

Figure 69: OEM Models

Model Template Package List

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”
selection when editing the model template.

• Click on to download the template.

• Click on to upgrade the model template. Users will see this icon available if the device model has

template updated in the UCM6200.

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 Figure 70: Template Management

Upload Model Template Package

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 packages can be manually
uploaded to the UCM.. Please contact Grandstream customer support if the model package is needed for manual
uploading.

Figure 71: Upload Model Template Manually

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

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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.

Figure 72: Create New Device

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.

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 Figure 73: 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.

Figure 74: Edit Device

A preview of the “Advanced” settings is shown in the above figure. There are five levels configurations as
described in (1) (2) (3) (4) (5) below, with priority from the lowest to the highest. The configurations in all levels
will take effect for the device. If there are same options existing in different level configurations with different
value configured, the higher-level configuration will override the lower level configuration.

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(1) Global Policy
This is the lowest level configuration. The global policy configured in Web 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 31: 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 75: Edit Customize Device Settings

Scroll down in the dialog to view and edit the device-specific options. If the users would like to add more options
which are not in the pre-defined list, click on “Add New Field” to add a P value number and the value to the
configuration. The above figure shows setting P value “P1362” to “en”, which means the display language on the
LCD is set to English. The warning information on right tells that the option matching the P value number exists
and clicking on it will lead to the matching option. For P value information of different models, please refer to
configuration template here http://www.grandstream.com/sites/default/files/Resources/config-template.zip.

• Select multiple devices that need to be modified and then click on to batch modify
devices.

If selected devices are of the same model, the configuration dialog is like the following figure. Configurations in
five levels are all available for users to modify.

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 Figure 76: Modify Selected Devices - Same Model

If selected devices are of different models, the configuration dialog is like the following figure. Click on to
view more devices of other models. Users are only allowed to make modifications in Global Templates and
Global Policy level.

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 Figure 77: Modify Selected Devices - Different Models

-------------------------------------------------------------------------------------------------------------------------------------------- -------

Note:
Performing batch operation will override all the existing device configuration on the page.
----------------------------------------------------------------------------------------------------------------------------- ----------------------

After the above configurations, save the changes and go back to Web 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 78: Device List in Zero Config

In this web page, users can also click on “Reset All Extensions” to reset the extensions of all the devices.

Example Application

Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1
GXV3370 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 79: Zero Config Sample - Global Policy

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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”.

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 80: Zero Config Sample - Device Preview 1

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8. For the 3 phones used by Spanish support, in “Basic” settings select an available extension for account 1
and click on “Save”. Then click on “Advanced” settings tab to bring up the following dialog.

Figure 81: 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 GXV3370 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 GXV3370.

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 Figure 82: Zero Config Sample - Device Preview 3

10. Click on “Apply Changes” to apply saved changes.

11. On the Web GUI→Value-added Features→Zero Config→Zero Config page, click on to send NOTIFY
to trigger the device to download config file from UCM6200.

Now all the 9 phones in the network will be provisioned with a 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 GXV3370 used by the supervisor will be provisioned to use the default language on LCD display
since it is not specified in the global policy.

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 EXTENSIONS

Create New User

Create New SIP Extension

To create a new SIP extension, navigate to Extension/Trunk->Extensions and click on the Add button. The
following window will appear:

Figure 83: Create New Device

Extension options are divided into four categories:

• Basic Settings
• Media
• Features
• Specific Time
• Follow me

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Select first which type of extension: SIP Extension, IAX Extension or FXS Extension. The configuration
parameters are as follows.
Table 32: SIP Extension Configuration Parameters – Basic Settings

General
The extension number associated with the user.
Extension
Note: This filed supports (+) sign.
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
Permission the highest level. The default setting is “Internal”.
Note: Users need to have the same level as or higher level than an
outbound rule’s privilege to make outbound calls using this rule.
Configure the password for the user. A random secure password will be
SIP/IAX Password automatically generated. It is recommended to use this password for
security purpose.
Configure the authentication ID for the user. If not configured, the
Auth ID extension number will be used for authentication.
Note: This filed supports (+) sign.
Configure Voicemail. There are three valid options and the default option
is "Enable Local Voicemail".
• Disable Voicemail: Disable Voicemail.
• Enable Local Voicemail: Enable voicemail for the user.
Voicemail
• Enable Remote Voicemail: Forward the notify message from
remote voicemail system for the user, and the local voicemail will
be disabled. Note: Remote voicemail feature is used only for
Infomatec (Brazil).
Configure the password to access the extension's voicemail. A randomly
Voicemail Password generated password is used by default and is highly recommended for
security. Only digits are supported.
Skip Voicemail Password If enabled, users can skip password verification when dialing in via the My
Verification Voicemail feature code. This option is disabled by default.
If enabled, the voicemail will be attached to the email notification that is
sent when a voicemail is received. Note: When set to "Default", the global
Send Voicemail to Email
settings in Call Features->Voicemail->Voicemail Email Settings will be
used.

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Keep Voicemail after Configure whether or not to retain the voicemail in local storage after
Emailing sending the voicemail attachment in the email notification.
If enabled, empty SDP packet will be sent to the SIP server periodically to
Enable Keep-alive
keep the NAT port open. The default setting is “Yes”.
Configure the Keep-alive interval (in seconds) to check if the host is up.
Keep-alive Frequency
The default setting is 60 seconds.
If selected, this extension will be disabled on the UCM6200.
Disable This Extension Note: The disabled extension still exists on the PBX but cannot be used
on the end device.
User Settings
Configure the user's first name. This field supports alphanumeric
First Name
characters, underscores (_), and periods.
Configures the user's last name. This field supports alphanumeric
Last Name
characters, underscores (_), and periods.
Configure the user's email address. Email notifications will be sent to this
Email Address
address.
Configure the password for user portal access. A random password is
User Password automatically generated by default and is highly recommended for
security.
Select the voice prompt language that will be used for this extension. By
default, the selected voice prompt language under PBX Settings->Voice
Language Prompt->Language Settings will be used. To add more supported
languages, please download the voice prompt language packages in the
same page.
The maximum allowed number of endpoint registrations to this extension.
Concurrent Registrations
Default value is 1.
Configure a phone number for the extension. Users can dial the Direct
Dial Mobile Phone Prefix feature code (*88 by default) + extension
Mobile Phone Number
number to directly dial this number. Example: Dial *881000 to call the
mobile number associated with extension 1000.

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 Table 33: 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
NAT
issue, usually it is related to NAT configuration or Firewall’s support of SIP
and RTP ports. The default setting is enabled.
By default, the UCM6200 will route the media steams from SIP endpoints
through itself. If enabled, the PBX will attempt to negotiate with the
Enable Direct Media 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
“RFC4733”. If “Info” is selected, SIP INFO message will be used. If
DTMF Mode “Inband” is selected, 64-kbit PCMU and PCMA are required. When “Auto”
is selected, RFC4733 will be used if offered, otherwise “Inband” will be
used.
If the phone has an assigned PSTN telephone number, this field should
be set to “User=Phone”. “User=Phone” parameter will be attached to the
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.
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.
Enable T.38 UDPTL Enable or disable T.38 UDPTL support.
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
Fax Mode 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.

• 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.

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 Access Control List manages the IP addresses that can register to this
extension.
ACL Policy • Allow All: Any IP address can register to this extension.
• Local Network Address: Only IP addresses in the configured
network segments can register to this extension.
Specifies allowed IP address or networks from where the extension can
Local Network Address be registered. Up to 10 entries are allowed.
Format: "xxx.xxx.xxx.xxx", "xxx.xxx.xxx.xxx/32", "[::]" or "[::]/128".
Select audio and video codec for the extension. The available codecs are:
Codec Preference PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723,
OPUS, iLBC, ADPCM, H.264, H.265, H.263, H.263p and RTX.

Table 34: SIP Extension Configuration Parameters – Features

Call Transfer
Select which presence status to set for the extension and configure call
forward conditions for each status. Six possible options are possible:
Presence Status
“Available”, “Away”, “Chat”, “Custom”, “DND” and “Unavailable”.
More details at [PRESENCE].
Enable and configure the Call Forward Unconditional target number.
Available options for target number are:
• “None”: Call forward deactivated.
• “Extension”: Select an extension from dropdown list as CFU target.
• “Custom Number”: Enter a customer number as target. For
example: *97.
Call Forward Unconditional • “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
CFU Time Condition Office Time or Holiday” and “Specific”.

Notes:
• “Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

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 • Specific Time can be configured in the Specific Time tab.
• Office Time and Holiday can be configured in System Settings->Time
Settings->Office Time/Holiday page.
Configure the Call Forward No Answer target number. Available options
for target number are:
• “None”: Call forward deactivated.
• “Extension”: Select an extension from dropdown list as CFN target.
• “Custom Number”: Enter a customer number as target. For
example: *97.
Call Forward No Answer
• “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.
Select time condition for Call Forward No Answer. The available time
conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday” and “Specific”.

Notes:
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 in the Specific Time tab.
• Office Time and Holiday can be configured in System Settings→Time
Settings→Office Time/Holiday page.
Configure the Call Forward Busy target number. Available options for
target number are:
• “None”: Call forward deactivated.
• “Extension”: Select an extension from dropdown list as CFB target.
• “Custom Number”: Enter a customer number as target. For
example: *97.
Call Forward Busy
• “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.
CFB Time Condition Select time condition for Call Forward Busy. The available time conditions

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 are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Notes:
• “Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.
• Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for
specific time.
• Office Time and Holiday could be configured on page System
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”.

Notes:
• “Specific” has higher priority to “Office Times” if there is a conflict
DND Time Condition 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.
If DND is enabled, all calls to this extension will be rejected except the
DND Whitelist numbers listed on this list.
Note: The maximum number on the Whitelist is 10.
If call forward is enabled, all calls to this extension will be forwarded except
FWD Whitelist the calls coming from the specified numbers on this list.
Note: The Maximum number on the whitelist is 10.
CC Settings

If enabled, UCM6200 will automatically alert this extension when a called

Enable CC party is available, given that a previous call to that party failed for some
reason. By default, it is disabled.

Two modes for Call Completion are supported:

• Normal: This extension is used as ordinary extension.

• For Trunk: This extension is registered from a PBX.
CC Mode
The default setting is “Normal”.

Note: The number of CC agents (for “Normal” mode) will now be limited
by the extensions’ allowed number of concurrent registrations.

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 Configure the maximum number of CCSS agents which may be allocated
for this channel (when CC Mode is set to “For Trunk”). In other words, this
CC Max Agents
number serves as the maximum number of CC requests this channel is
allowed to make. Min. value is 1.
Configure the maximum number of monitor structures which may be
created for this device (when CC Mode is set to “For Trunk”). This number
CC Max Monitors
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 extension. If a register trunk is used for outbound, the register
Ring Simultaneously
number will be used to be displayed for the external number as caller ID
number.
Set the external number to be rang simultaneously. ‘-’ is the connection
External Number character which will be ignored.
This field accepts only letters, numbers, and special characters + = * #.
Time Condition for Ring Ring the external number simultaneously along with the extension based
Simultaneously on this time condition.
Use callee DOD on FWD or Use the DOD number when calls are being diverted/forwarded to external
Ring Simultaneously destinations or when ring simultaneous is configured.
Monitor privilege control
Add members from “Available Extensions” to “Selected Extensions” so
Allow call-barging that the 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
Allowed to seamless transfer Pickup Incall 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 global ring timeout setting under
Ring Timeout Web GUI→PBX Settings→General Settings: 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.

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 Enable automatic recording for the calls using this extension. The default
Auto Record setting is 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
Skip Trunk Auth
outbound calls during the selected time condition.
• If set to ‘No’, users will be asked to enter the password when making
outbound calls.
Time Condition for Skip If ‘Skip Trunk Auth’ is set to ‘By Time’, select a time condition during which
Trunk Auth users can skip entering password when making outbound calls.
Dial Trunk Password Configure personal password when making outbound calls via trunk.
If enabled, SIP Password will accept only alphabet characters and digits.
Support Hot-Desking Mode
Auth ID will be changed to the same as Extension.
Enable LDAP If enabled, the extension will be added to LDAP Phonebook PBX list.
Enable WebRTC Support Enable registration and call from Web RTC.
Specify which Music On Hold class to suggest to the bridged channel when
Music On Hold
putting them on hold.
Enable Seamless Transfer Enable the seamless transfer for this extension.
Permission Set the permission for this extension when using the seamless transfer
Call Duration Limit The maximum duration of call-blocking.
The maximum call duration (in seconds). The default value 0 means no
Maximum Call Duration
limit.

Custom Call-info for Auto If enabled, when a call is sent to this extension from UCM, the SIP
Answer INVITE message will contain a Call-info header indicating auto answer.

If disabled, UCM will not ring the extension if it is already in a call and will
Enable Call Waiting
give the caller a busy response.

Table 35: 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.

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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 36: 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
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
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.
Configure the password for the user. A random secure password will be automatically
SIP/IAX Password
generated. It is recommended to use this password for security purpose.
Configure Voicemail. There are three valid options and the default option is "Enable
Local Voicemail".
Voicemail
• Disable Voicemail: Disable Voicemail.
• Enable Local Voicemail: Enable voicemail for the user.
Configure the password to access the extension's voicemail. A randomly generated
Voicemail
password is used by default and is highly recommended for security. Only digits are
Password
supported.
Skip Voicemail
If enabled, users can skip password verification when dialing in via the My Voicemail
Password
feature code. This option is disabled by default.
Verification
If selected, this extension will be disabled on the UCM6200.
Disable This
Note:
Extension
The disabled extension still exists on the PBX but cannot be used on the end device.
User Settings
Configure the user's first name. This field supports alphanumeric characters,
First Name
underscores (_), and periods.
Configures the user's last name. This field supports alphanumeric characters,
Last Name
underscores (_), and periods.
Email Address Configure the user's email address. Email notifications will be sent to this address.
Configure the password for user portal access. A random password is automatically
User Password
generated by default and is highly recommended for security.

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 Select the voice prompt language that will be used for this extension. By default, the
selected voice prompt language under PBX Settings->Voice Prompt->Language
Language
Settings will be used. To add more supported languages, please download the voice
prompt language packages in the same page.

Table 37: IAX Extension Configuration Parameters→Media

IAX Settings
Max Number of
Configure the maximum number of calls allowed for each remote IP address.
Calls
Configure to enable/disable requiring call token. If set to "Auto", it might lock out users
Require Call
who depend on backward compatibility when peer authentication credentials are
Token
shared between physical endpoints. The default setting is "Yes".
SRTP 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.
Access Control List manages the IP addresses that can register to this extension.
• Allow All: Any IP address can register to this extension.
ACL Policy
Local Network Address: Only IP addresses in the configured network segments can
register to this extension.
Select audio and video codec for the extension. The available codecs are: PCMU,
Codec Preference PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, iLBC, ADPCM, H.264,
H.265, H.263, H.263p, RTX and VP8.

Table 38: 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
CFU Time
Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday” and “Specific”.
Condition
Notes:
• “Specific” has higher priority to “Office Times” if there is a conflict in terms of time

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 period.
• Specific Time can be configured in the Specific Time tab.
• Office Time and Holiday can be configured in 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”.
Notes:
CFN Time
• “Specific” has higher priority to “Office Times” if there is a conflict in terms of time
Condition
period.
• Specific Time can be configured in the Specific Time tab.
• Office Time and Holiday can be configured in System Settings→Time
Settings→Office Time/Holiday page.
Configure the Call Forward Busy target number. If not configured, the Call Forward
Call Forward Busy
Busy feature is deactivated. The default setting is deactivated.
Select time condition for Call Forward Busy. The available time conditions are “Office
Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”
and “Specific”.
Notes:
CFB 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.
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.
Set the external number to be rang simultaneously. ‘-’ is the connection character
External Number
which will be ignored.
Time Condition for
Ring the external number simultaneously along with the extension on the basis of this
Ring
time condition.
Simultaneously
Other Settings
Configure the number of seconds to ring the user before the call is forwarded to
Ring Timeout
voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified,

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 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 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
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 39: 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 40: FXS Extension Configuration Parameters→Basic Settings

General
Extension The extension number associated with the user.

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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
user.
Caller ID 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.
Configure Voicemail. There are three valid options and the default option is "Enable
Local Voicemail".
Voicemail
• Disable Voicemail: Disable Voicemail.
• Enable Local Voicemail: Enable voicemail for the user.
Configure the password to access the extension's voicemail. A randomly generated
Voicemail
password is used by default and is highly recommended for security. Only digits are
Password
supported.
Skip Voicemail
If enabled, users can skip password verification when dialing in via the My Voicemail
Password
feature code. This option is disabled by default.
Verification
If selected, this extension will be disabled on the UCM6200.
Disable This
Note: The disabled extension still exists on the PBX but cannot be used on the end
Extension
device.
User Settings
Configure the user's first name. This field supports alphanumeric characters,
First Name
underscores (_), and periods.
Configures the user's last name. This field supports alphanumeric characters,
Last Name
underscores (_), and periods.
Email Address Configure the user's email address. Email notifications will be sent to this address.
Configure the password for user portal access. A random password is automatically
User Password
generated by default and is highly recommended for security.
Select the voice prompt language that will be used for this extension. By default, the
selected voice prompt language under PBX Settings->Voice Prompt->Language
Language
Settings will be used. To add more supported languages, please download the voice
prompt language packages in the same page.

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 Table 41: FXS Extension Configuration Parameters→Media

Analog Settings
Configure to enable/disable call waiting feature.
Call Waiting
The default setting is "No".
If configured, the # key can be used as SEND key after dialing the number on the
User ‘#’ as SEND
analog phone. The default setting is "Yes".
Configure the RX gain for the receiving channel of analog FXS port. The valid range
RX Gain
is -30dB to +6dB. The default setting is 0.
Configure the TX gain for the transmitting channel of analog FXS port. The valid range
TX Gain
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
remain unpressed for the PBX to consider the event as a valid flash event. The
MAX RX Flash
minimum period of time is 256ms and it cannot be modified. The default setting is
1250ms.
If enabled, a polarity reversal will be marked as received when an outgoing call is
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.
Configure to enable/disable 3-way calling feature on the user. The default setting is
3-Way Calling
enabled.
Send CallerID
Configure the number of rings before sending CID. Default setting is 1.
After
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
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.
• Fax Gateway: If selected, the UCM6200 can support conversation and

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 processing of Fax data from T.30 to T.38 or T.38 to T.30. only for FXS ports.

Table 42: 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”.
Notes:
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 in the Specific Time tab.
• Office Time and Holiday can be configured in 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”.
Notes:
CFN Time
• “Specific” has higher priority to “Office Times” if there is a conflict in terms of time
Condition
period.
• Specific Time can be configured in the Specific Time tab.
• Office Time and Holiday can be configured in System Settings→Time
Settings→Office Time/Holiday page.
Configure the Call Forward Busy target number. If not configured, the Call Forward
Call Forward Busy
Busy feature is deactivated. The default setting is deactivated.
Select time condition for Call Forward Busy. The available time conditions are “Office
Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of Office Time or Holiday”
and “Specific”.
Notes:
CFB 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.

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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.

Set the external number to be rang simultaneously. ‘-’ is the connection character
External Number
which will be ignored.

Time Condition for

Ring the external number simultaneously along with the extension on the basis of this
Ring
time condition.
Simultaneously
Hotline

If enabled, hotline dialing plan will be activated, a pre-configured number will be

Enable Hotline
used based on 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 global ring timeout setting under Web GUI→PBX Settings→Voice
Ring Timeout
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.

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 • 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.

Table 43: 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 create multiple SIP extensions quickly, users can select "Batch" for the Select Add Method option during the
extension creation process.
Table 44: Batch Add SIP Extension Parameters

General
Create Number Specify the number of extensions to be added. The default setting is 5.
Extension Specify how much to increment each additional extension (e.g., if the value is 2, the
Incrementation extensions will be 1000, 1002, 1004, etc.). Note: Supports up to 3 characters.
Extension Configure the starting extension number of the batch of extensions to be added.
Assign permission level to the user. The available permissions are "Internal", "Local",
"National" and "International" from the lowest level to the highest level. The default
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.
Configure Voicemail. There are three valid options and the default option is "Enable
Voicemail Local Voicemail".
• Disable Voicemail: Disable Voicemail.

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 • Enable Local Voicemail: Enable voicemail for the user.
• Enable Remote Voicemail: Forward the notify message from remote
voicemail system for the user, and the local voicemail will be disabled. Note:
Remote voicemail feature is used only for Infomatec (Brazil).
Enable WebRTC
Enable WebRTC support.
Support
Configure the SIP/IAX password for the users. Three options are available to create
password for the batch of extensions.
• User Random Password.
SIP/IAX 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 the password to access the extension's voicemail. A randomly generated
Voicemail
password is used by default and is highly recommended for security. Only digits are
Password
supported.
Configure CallerID Number when adding Batch Extensions.

• Use Extension as Number

CallerID 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
If enabled, users can skip password verification when dialing in via the My Voicemail
Password
feature code. This option is disabled by default.
Verification
Select which Music On Hold class to suggest to extensions when putting them on
Music On Hold
hold.

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 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 (e.g., broadband router). If there is one-way audio issue, usually it is related to
NAT
NAT configuration or Firewall's support of SIP and RTP ports.
The default setting is enabled.

By default, the PBX will route the media steams from SIP endpoints through itself. If
Enable Direct enabled, the PBX will attempt to negotiate with the endpoints to route the media
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 "RFC4733". 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, RFC4733 will be
used if offered, otherwise "Inband" will be used.

If enabled, empty SDP packet will be sent to the SIP server periodically to keep the
Enable Keep-alive
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 allowed number of endpoint registrations to this extension. Default
Registrations value is 1.

Other Settings
SRTP Enable SRTP for the call. The default setting is "No".
Fax Mode Select Fax mode for this user. The default setting is “None”.

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 • None: Disable Fax.

• Fax Detect: Fax signal from the user/trunk during the call can be detected and
the received Fax will be sent to the Email address configured for this extension.
If no Email address can be found for the user, the Fax will be sent to the default
Email address configured in Fax setting page under Web GUI→Call
Features→Fax/T.38.
Access Control List manages the IP addresses that can register to this extension.
• Allow All: Any IP address can register to this extension.
ACL Policy
Local Network Address: Only IP addresses in the configured network segments
can register to 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.265, 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 45: 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.
Configure Voicemail. There are three valid options and the default option is "Enable
Local Voicemail".
Voicemail
• Disable Voicemail: Disable Voicemail.
• Enable Local Voicemail: Enable voicemail for the user.
Configure the SIP/IAX password for the users. Three options are available to create
password for the batch of extensions.
SIP/IAX Password
• User Random Password.
A random secure password will be automatically generated. It is recommended

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 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 the password to access the extension's voicemail. A randomly generated
Voicemail
password is used by default and is highly recommended for security. Only digits are
Password
supported.
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
If enabled, users can skip password verification when dialing in via the My Voicemail
Password
feature code. This option is disabled by default.
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
users who depend on backward compatibility when peer authentication credentials
Require Call Token
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

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 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.
Access Control List manages the IP addresses that can register to this extension.
• Allow All: Any IP address can register to this extension.
ACL Policy
Local Network Address: Only IP addresses in the configured network segments
can register to this extension.
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.265, H.263, H.263p and VP8.

Batch Extension Resetting Functionality

To reset multiple extensions to default settings, select the extensions on the main Extensions/Trunks-
>Extensions page and click on the Reset button. Once reset, all settings except for Concurrent Registrations
will be reverted to default. User voicemail password will be randomized again, and user voicemail prompts and
recordings will be deleted. The settings for the extension user in the User Management page will also be
reverted to default except for usernames and custom privileges.

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 "Edit
Selected Extensions" or "Delete Selected Extensions". Also, options "Edit” , "Reboot" , "Delete" and
reset are available per extension. User can search an extension by specifying the extension number to find an
extension quickly.

Figure 84: Manage Extensions

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• Status
Users can see the following icon for each extension to indicate the SIP status.
Green: Idle
Blue: Ringing
Yellow: In Use
Grey: Unavailable (the extension is not registered or disabled on the PBX)

• Edit single extension

Click on to start editing the extension parameters.

• Reset single extension

Click on to reset the extension parameters to default (except concurrent registration).

Other settings will be restored to default in Maintenance→User Management→User Information except

username and permissions and delete the user voicemail prompt and voice messages.

• Reboot the user

Click on to send NOTIFY reboot event to the device which has an 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 selected extensions

Click on the button to delete a single extension. Users can also select multiple extensions and click on
the Delete button to delete several extensions at once.

• Modify selected extensions

Select the checkbox for the extension(s). Then click on "Edit" to edit the extensions in a batch.

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Export Extensions

UCM extensions can be exported to a CSV file. Different extension types (SIP, IAX, FXS) cannot be exported to
the same CSV file. To export extensions, click on the Export button and select the extension type to export.

Figure 85: Export Extensions

Users can also use the exported CSV file to use as a template to manually edit extension information.

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.

• Export extension csv file from the UCM6200 by clicking on "Export Extensions" button.
• Fill up the extension information you would like in the exported csv template.
• Click on "Import Extensions" button. The following dialog will be prompted.

Figure 86: Import Extensions

• 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.

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 • 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.
• Click on “Choose file to upload” to select csv file from local directory in the PC.
• Click on "Apply Changes" to apply the imported file on the UCM6200.

Example of file to import:

Figure 87: Import File

Table 46: SIP extensions Imported File Example

Field Supported values

Extension Digits
Technology SIP/SIP(WebRTC)
Enable Voicemail yes/no/remote
CallerID Number Digits
SIP/IAX Password Alphanumeric characters
Voicemail Password Digits
Skip Voicemail Password Verification yes/no
Ring Timeout Empty/ 3 to 600 (in second)
SRTP yes/no
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

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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,
H.265,ILBC,AAL2-G.726-
32,ADPCM,G.723,H.263,H.263p,vp8,opus
Permission Internal/Local/National/International
NAT yes/no
DTMF Mode RFC4733/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
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

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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
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 47: 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

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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,
H.265,ILBC,AAL2-G.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

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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

Table 48: 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

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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
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

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 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 88: Import Error

Extension Details

Users can click on an extension number in the Extensions list page and quickly view information about it such
as:
• Extension: Shows the Extension number.
• Status: Shows the status of the extension.
• Presence status: Indicates the Presence Status of this extension.
• Terminal Type: Shows the Type of the terminal using this extension (SIP, FXS…etc.).
• Caller ID Name: Reveals the Caller ID Name configured on the extension.
• Messages: Shows the messages stats.
• IP and Port: The IP address and the ports of the device using the extension.
• Email status: Show the Email status (sent, to be sent…etc.).
• Ring Group: Indicates the ring groups that this extension belongs to.
• Call Queue: Indicates the Cal Queues that this extension belongs to.
• Call Queue (Dynamic): Indicates the Call Queues that this extension belongs to as a dynamic agent.

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 Figure 89: Extension Details

E-mail Notification

Users can request for their SIP acccount information via email. The system administrator can select the
extensions to send account information emails to and press the Email Notification button in the Extensions
page.

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. After clicking the button, the following
prompt will appear:

Figure 90: E-mail Notification - Prompt Information

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The selected users will receive emails containing account registration information, LDAP configuration, and QR
codes for quick setup on mobile apps.

Figure 91: Account Registration Information and QR Code

Figure 92: LDAP Client Information and QR Code

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Multiple Registrations per Extension

UCM6200 supports multiple registrations per extension so that users can use the same extension on devices in
different locations.

Figure 93: 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 1 for security purposes. Maximum is 10.

Figure 94: 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 95: 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 to create a new extension group.

• Click on to edit the extension group.

• Click on to delete the extension group.

Select extensions from the list on the left side to the right side.

Figure 96: Edit Extension Group

Click on in order to change the ringing priority of the members selected on the 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 97: 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 49: Analog Trunk Configuration Parameters

Select the channel for the analog trunk.

• UCM6202: 2 channels
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.
Advanced Options
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.
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 “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 -12.0 (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 -12.0 (dB) to + 12.0 (dB). The default setting is 0.
Use CallerID Configure to enable CallerID detection. The default setting is “Yes”.
Select the Caller ID scheme for this trunk.
• Bellcore/Telcordia.
• ETSI-FSK During Ringing
• ETSI-FSK Prior to Ringing with DTAS
• ETSI-FSK Prior to Ringing with LR
• ETSI-FSK Prior to Ringing with RP
• ETSI-DTMF During Ringing
Caller ID Scheme • ETSI-DTMF Prior to Ringing with DTAS
• ETSI-DTMF Prior to Ringing with LR
• ETSI-DTMF Prior to Ringing with RP
• SIN 227-BT
• NTT Japan
• Auto Detect
If you are not sure which scheme to choose, please select “Auto Detect”. The default
setting is “Bellcore/Telcordia”.
Enable to detect Fax signal from the trunk during the call and send the received Fax
to the default Email address in Fax setting page under Web GUI→Call
Fax Mode
Features→Fax/T.38. The default setting is “No”.
Note: If enabled, Fax Pass-through cannot be used.
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.

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 If selected, the trunk will be disabled and incoming/Outgoing calls via this trunk will
Disable This Trunk
not be possible.
This is to implement analog trunk outbound line selection strategy.
Three options are available:

• 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 is 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 is idle).

The default setting is “Ascend” mode.

The Non-Linear Processing (NLP) in echo cancellation helps to remove/suppress

residual echo components that could not be removed by the LEC (Line Echo
Canceller). Following modes are supported:

• Default: Slightly reduces the amount of detected background noise.

• High Noise Level Adjustment: Greatly reduces the amount of detected

Echo Cancellation
background noise.
Mode
• Noise Masking: Injects noise to mask echo and greatly reduces the amount of
detected background noise.

• White Noise Injection: Injects variable amounts of white noise based on

detected background noise and greatly reduces the amount of detected
background noise.

Allows external numbers the option to get directed to the extension that last called
them.

For Example: User 2002 has dialed external number 061234575 but they were not
Direct Callback
reachable, once they have received missed call notification on their phone, they
would mostly call back the number, if the option “Direct Callback” is enabled then
they will be directly bridged to user 2002 regardless of the configured inbound
destination.

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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".
If "Congestion Detection" is enabled, users can specify the number of congestion
Congestion Count
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
by PSTN. Before the detecting, please make sure there is more than one channel
PSTN Detection
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 98: UCM6200 FXO Tone Settings

4. Click on "Detect" to start PSTN detection.

Figure 99: 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 100: UCM6200 PSTN Detection: Auto Detect

• If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.

Figure 101: 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 50: 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.

Note: UCM6200 supports now 200 VoIP trunks

• 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/ucm6xxx_sip_trunk_guide.pdf

The VoIP trunk options are listed in the table below.

Table 51: Create New SIP Trunk

Select the VoIP trunk type.

Type • Peer SIP Trunk
• Register SIP Trunk

Configure a unique label (up to 64 character) to identify this trunk when listed in
Provider Name
outbound rules, 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 ensure that the remote peer PBX supports
matching user entry via the "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”.

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 Turn on this setting when the PBX is using public IP and communicating with
NAT devices behind NAT. If there is one-way audio issue, usually it is related to NAT
configuration or SIP/RTP port support on the firewall.

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.

Important Note: When making outgoing calls, the following priority order rule will
Caller ID be used to determine which CallerID will be set before sending out the call:

From user (Register Trunk Only) → CID from inbound call (Keep Original CID
Enabled) → Trunk Username/CallerID (Keep Trunk CID Enabled) → DOD →
Extension CallerID Number → Trunk Username/CallerID (Keep Trunk CID
Disabled) → Global Outbound CID.
Select whether the trunk needs to register on the external server or not when
Need Registration
"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
Username
Trunk" type is selected.
Enter the password to register to the trunk from the provider when "Register SIP
Password
Trunk" is selected.
Auth ID 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.
Allows external numbers the option to get directed to the extension that last called
them.
For Example: User 2002 has dialed external number 061234575 but they were not
Direct Callback reachable, once they have received missed call notification on their phone, they
would mostly call back the number, if the option “Direct Callback” is enabled then
they will be directly bridged to user 2002 regardless of the configured inbound
destination.

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 Table 52: SIP Register 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.
Configure the SIP transport protocol to be used in this trunk. The default setting is
"UDP".
Transport • UDP
• TCP
• TLS
SIP URI Scheme When TLS is selected as Transport for register trunk, users can select between
When Using TLS SIP and SIPS URI scheme
Keep the CID from the inbound call when dialing out. This setting will override
Keep Original CID “Keep Trunk CID” option. Please ensure that the remote peer PBX supports
matching user entry via the "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 is 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.
Select whether the trunk needs to register on the external server or not when
Need Registration
"Register SIP Trunk" type is selected. The default setting is No.
Allow outgoing calls If enabled, outbound calls can still be made even if registration to the trunk fails.
if registration failure Note: If Need Registration is disabled, this option will be ignored.
Configure the new name of the caller when the extension has no CallerID Name
CallerID Name
configured.
Configure the domain name of the SIP user account. This will overwrite the
From Domain
domain in the "From" header. Example: If the user account is "1234567", setting

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 "trunk.UCM6510.provider.com" as the From Domain value will result in the
following From header: "sip:1234567@trunk.UCM6510.provider.com".
Configure the username of the account. This will overwrite the user value in the
From header. This is useful for scenarios where a single user account has
From User multiple DIDs. Example: If the domain is "trunk.UCM6510.provider.com", setting
"1234567" as the From User value will result in the following From header:
"sip:1234567@trunk.UCM6510.provider.com".
Enter the username to register to the trunk from the provider when "Register SIP
Username
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.
If enabled, the UCM will send 401 response to the incoming call to authenticate
Auth Trunk
the trunk.
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.
Allows external numbers the option to get directed to the extension that last called
them.
For Example: User 2002 has dialed external number 061234575 but they were not
Direct Callback reachable, once they have received missed call notification on their phone, they
would mostly call back the number, if the option “Direct Callback” is enabled then
they will be directly bridged to user 2002 regardless of the configured inbound
destination.
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.265, H.263, H.263p, RTX, OPUS and VP8.

If enabled, the SIP INVITE message sent to the trunk will contain PPI (P-Preferred-
Identity) 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-Asserted-
Send PAI Header
Identity) header including configured PAI Header. The default setting is “No”.

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 Note: “Send PPI Header” and “Send PAI Header” cannot be enabled at the same
time. Only one of the two headers can be contained in the SIP INVITE message.
If “Send PAI Header” is enabled and “PAI Header” is configured as “123456” for
instance, the PAI header in the SIP message sent from the UCM will contain
“123456”. If “Send PAI Header” is enabled and “PAI Header” is configured as
“empty”, the PAI header in the SIP message sent from the UCM will contain the
PAI Header
original CID.
Note: “Send PAI Header” needs to be enabled in order to use this feature.
Only alphanumeric characters are allowed and/or special characters #*-_+. with a
limit of 64 characters.
If enabled and "From User" is configured, the INVITE's From header will contain
DOD As From Name the DOD number.

Passthrough PAI If checked and option "Send PAI Header" not checked, the PAI header will be
Header passthrough from one side to the other side.

Outbound Proxy
Select to enable outbound proxy in this trunk. The default setting is "No".
Support
When outbound proxy support is enabled, enter the IP address or URL of the
Outbound Proxy
outbound proxy.
It is used to set if the phone system will remove outbound proxy URI from the route
Remove OBP from
header. If is set to “Yes”, it will remove the route header from SIP requests. The
Route
default setting is “No”.
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.

• RFC4733: Send DTMF using RFC4733.

DTMF Mode
• 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 RFC4733 if offered. Otherwise, inband will be

used.
If enabled, the UCM6200 will regularly send SIP OPTIONS to the trunk to check
Enable Heartbeat
connection status.
Detection
The default setting is "No".

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 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”.
• 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.
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.

Table 53: 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.
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.
Keep the CID from the inbound call when dialing out, this setting will override “Keep
Keep Original CID Trunk CID” option. Please ensure that the remote peer PBX supports matching
user entry via the "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”.
NAT Turn on this option when the PBX is using public IP and communicating with

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 devices behind NAT. If there is one-way audio issue, usually it is related to NAT
configuration or SIP/RTP port configuration on the firewall.
If selected, the trunk will be disabled.
Note:
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.
Important Note: When making outgoing calls, the following priority order rule will
Caller ID be used to determine which CallerID will be set before sending out the call :
• CID from inbound call (Keep Original CID Enabled) → Trunk
Username/CallerID (Keep Trunk CID Enabled) → DOD → Extension CallerID
Number → Trunk Username/CallerID (Keep Trunk CID Disabled) → Global
Outbound CID.
Configure the name of the caller to be displayed when the extension has no
CallerID Name
CallerID Name configured.
Configure the SIP transport protocol to be used in this trunk. The default setting is
"UDP".
Transport • UDP
• TCP
• TLS
Allows external numbers the option to get directed to the extension that last called
them.
For Example: User 2002 has dialed external number 061234575 but they were not
Direct Callback reachable, once they have received missed call notification on their phone, they
would mostly call back the number, if the option “Direct Callback” is enabled then
they will be directly bridged to user 2002 regardless of the configured inbound
destination.
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.265, H.263, H.263p,RTX, OPUS and VP8.
DID Mode Configure where to get the destination ID of an incoming SIP call, from SIP

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 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 • RFC4733: Send DTMF using RFC4733.

• Info: Send DTMF using SIP INFO message.

• Inband: Send DTMF using inband audio. This requires 64-bit codec, i.e.,
PCMU and PCMA.

• Auto: Send DTMF using RFC4733 if offered. Otherwise, inband is used.

Enable Heartbeat If enabled, the UCM6200 will regularly send SIP OPTIONS to the trunk to check
Detection connection status. 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”.
• 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.
Enable SRTP for the VoIP trunk.
SRTP
The default setting is "No".
If enabled, it will allow SDP passthrough to Grandstream IPVideoTalk therefore it
IPVT Mode
will allow calls between the UCM and IPVideoTalk. The default setting is disabled.
If enabled, the local UCM6200 will automatically provide and update the local
LDAP contacts to the remote UCM6200 SIP peer trunk. In order to ensure
Sync LDAP Enable successful synchronization, the remote UCM6200 peer also needs to enable this
option on the SIP peer trunk.
The default setting is "No".
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.

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 Specify an outbound rule for LDAP sync feature. The UCM6200 will automatically
LDAP Outbound Rule
modify the remote contacts by adding prefix parsed from this rule.
Specify the prefix for LDAP sync feature. The UCM6200 will automatically modify
LDAP Dialed Prefix
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
this device. In other words, this number tells how many callers may request CC
CC Max Monitors
services for a specific device at one time.
The minimum value is 1.

Table 54: 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 55: IAX Register 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".

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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.
Important Note: When making outgoing calls, the following priority order rule will
Caller ID be used to determine which CallerID will be set before sending out the call :
From user (Register Trunk Only) → CID from inbound call (Keep Original CID
Enabled) → Trunk Username/CallerID (Keep Trunk CID Enabled) → DOD →
Extension CallerID Number → Trunk Username/CallerID (Keep Trunk CID
Disabled) → Global Outbound CID.

Configure the name of the caller to be displayed when the extension has no
CallerID Name
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.265, 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”.

• 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.

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 Table 56: 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.

Important Note: When making outgoing calls, the following priority order rule will
Caller ID
be used to determine which CallerID will be set before sending out the call :

CID from inbound call (Keep Original CID Enabled) → Trunk Username/CallerID
(Keep Trunk CID Enabled) → DOD → Extension CallerID Number → Trunk
Username/CallerID (Keep Trunk CID Disabled) → Global Outbound CID..
Configure the name of the caller to be displayed when the extension has no
CallerID Name
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.265, 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”.
• 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.

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Trunk Groups

Users can create VoIP Trunk Groups to register and easily apply the same settings on multiple accounts within
the same SIP server. This can drastically reduce the amount of time needed to manage accounts for the same
server and improve the overall cleanliness of the web UI.

Figure 102: Trunk Group

Once creating the new trunk group and configuring the SIP settings, users can add multiple accounts within the

configured SIP server by pressing button and configuring the username, password and authentication ID

fields.

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 Figure 103: Trunk Group Configuration

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 to configure DOD on the UCM6200:

1. To setup DOD go to UCM6200 Web GUI→Extension/Trunk→VoIP Trunks page.

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2. Click to access the DOD options for the selected SIP Trunk.

3. Click "Add DOD" to begin your DOD setup

4. For "DOD Number" enter one of the numbers (DIDs) from your SIP trunk provider. In the example above
Company ABC received 4 DIDs from their provider. ABC will enter in the number for the CEO's direct line.
Note: DOD number cannot exceed 32 characters.

5. Set the DOD name and If extension number need to be appended to the DID number click on “Add
Extension”.
Note: DOD name cannot exceed 32 characters.

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.

Figure 104: DOD extension selection

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 105: Edit DOD

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DOD can also be assigned to the UCM’s Fax Sending feature. To do this, select “VFAX” from the extension list
when creating or editing a DOD number like shown below.

Figure 106: Fax Sending DOD

DOD Import/Export

UCM62xx now supports to import/export DOD CSV file for an easy configuration.

• To Import DOD CSV file, click on “Import”.

Note: the file must contain and respect the following columns: DOD Number, DOD Name, Add extension, Local
Member, LDAP Members.

Figure 107: DOD Import

• To export the DOD file, users can click on “Export”.

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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 108: SLA Station

• Click on to add an SLA Station.

• Click on to edit the SLA Station. The following table shows the SLA Station configuration parameters.

• Click on to delete the SLA Station.

Table 57: 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
to arrange the order. If there are multiple trunks selected, when there are
Selected SLA Trunks
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
Configure the time (in seconds) to ring the station before the call is considered
Ring Timeout
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 109: Enable SLA Mode for Analog Trunk

2. Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.

Figure 110: 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 111: 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 112: 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 an FXO while
"Long distance" 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to be used
when the primary trunk fails.

Note: UCM6200 supports now 500 Outbound routes.

Go to Web GUI→Extension/Trunk→Outbound Routes to add and edit outbound rules.

• Click on to add a new outbound route.

• Click on to edit the outbound route.

• Click on to delete the outbound route.

On the UCM6200, the outbound route priority is based on “Best matching pattern”. For example, the UCM6200
has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured. When dialing 1000
for outbound call, outbound route B will always be used first. This is because pattern 10xx is a better match than
pattern 1xxx. Only when there are multiple outbound routes with the same pattern configured.

Table 58: Outbound Route Configuration Parameters

Configure the name of the calling rule (e.g., local, long_distance, and etc.).
Calling Rule Name
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:

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 ▪ Multiple patterns can be used. Each pattern should be entered in
new line.
▪ Users can add comments to the end of patterns to better organize
and keep track of complex rules by typing “/*” and “*/” before and
after each comment, respectively.
▪ Example:
_X.
_NNXXNXXXXX /* 10-digit long distance */
_818X. /* Any number with leading 818 */
After creating the outbound route, users can choose to enable and disable
Disable This Route it. If the route is disabled, it will not take effect anymore. However, the route
settings will remain in UCM. Users can enable it again when it is needed.

Configure the password for users to use this rule when making outbound
Password
calls.

Once call duration limit is enabled, it will set the maximum duration of call-
Call Duration Limit
blocking.

User can customize the maximum call duration (in seconds) that is allowed
Maximum Call Duration for the outbound call. By default, this value is set to 0 means there is no
limit for the call duration.

This option will give caller warning when call duration is approaching to its
Warning Time limit. If the warning time is set to ‘y’, the warning tone will be played to caller
when y seconds is left to end the call by UCM.

Once this option is set to ‘z’, it will repeatedly be warning caller every z
Warning Repeat Interval
seconds after the first warning.

If selected, the Password, Privilege Level and Enable Filter on Source

PIN Groups Caller ID will not take effect.
For more details, refer to [PIN Groups] section.

PIN Groups with Privilege If enabled and PIN Groups are used, Privilege Levels and Filter on Source
Level Caller ID will also be applied.

Configure the password for users to use this rule when making outbound
Password
calls.

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 Select privilege level for the outbound rule.

• Internal:
The lowest level required. All users can use this rule.

• Local:
Users with Local, National, or International level are allowed to use this
rule.

• National:
Privilege Level
Users with National or International level are allowed to use this rule.

• International:
The highest level required. Only users with international level can use
this rule.

The default setting is "Disable". Please be aware of the potential security

risks when using "Internal" level, which means all users can use this
outbound rule to dial out from the trunk.

When enabled, users could specify extensions allowed to use this outbound
route. "Privilege Level" is automatically disabled if using "Enable Filter on
Source Caller ID".
The following two methods can be used at the same time to define the
extensions as the source caller ID.
1. Select available extensions/extension groups from the left to the right.
This allows users to specify arbitrary single extensions available in
the PBX.
2. Custom Dynamic Route: define the pattern for the source caller ID.
This allows users to define extension range instead of selecting them
Enable Filter on Source
one by one.
Caller ID
• 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.

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 Attempt to use the configured outbound route CID. This CID will not be used
Outbound Route CID if DOD is configured. It is formatted as "name<number>" or "<number>" or
"number".

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.
Prepend
Those digits will be prepended after the dialing number is stripped.

Use Failover Trunk

Failover trunks can be used to ensure calls can still be made in the scenario
that the main trunk is congested or down.
UCM6200 support up to 10 failover trunks.
Failover Trunk
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.
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.
Prepend
Those digits will be prepended after the dialing number is stripped.

Time Condition

Outbound routes can be configured to use different trunks based on time

Time Condition
conditions such as holidays, office times, specific times, etc.

Outbound Blacklist

The UCM6200 allows users to configure blacklist for outbound routes. If the dialing number matches the blacklist

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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 113: Country Codes

Blacklist Manage

Add Blacklist Rule Allows to define a rule based on number(s) or pattern(s) as blacklist entry.
Pattern rules:

N : Any digit from 2-9

X : Any digit from 0-9

Z : Any digit from 1-9

. : Wildcard, matching one or more characters

! : Wildcard, matching zero or more characters immediately

- : Hyphens are used mainly to improve readability and are not involved in
pattern matching.

Note: Users can export outbound route blacklists and delete all blacklist entries. Additionally, users can also

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import blacklists for outbound routes.

Figure 114: Blacklist Import/Export

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.

Table 59: Outbound Routes/PIN Group

Name Specify the name of the group

Specify Whether or not to display the used PIN group and name in a call's CDR
Record In CDR
entry.

PIN Number Specify the PIN that will be required to dial out.

PIN Name Specify the name of the PIN

Once user click on the following figure shows to configure the new PIN.

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 Figure 115: Create New PIN Group

The following screenshot shows an example of created PIN Groups and members:

Figure 116: PIN Members

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 Figure 117: 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.

Figure 118: 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.

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 Figure 119: Importing PIN Groups from CSV files

2. Select the CSV file to upload. Incorrect file formats and improperly formatted CSV files will result in
error messages such as the one below:

Figure 120: Incorrect CSV File

3. To ensure a successful import, please follow the format in the sample image below

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 Figure 121: CSV File Format

• The top-left value (A1) is the PIN Group name. In this case, it is “ALPHA”.
• Row 2 contains the labels for the modifiable fields: pin and pin_name. These values should not
be changed and will cause an upload error otherwise.

• Rows 3+ contain the user-defined values with Column A holding the PINs and Column B
holding the PIN names. PIN values must consist of at least four digits.
• Once the file is successfully uploaded, the entry will be added to the list of PIN Groups.

Figure 122: CSV File Successful Upload

- Exporting PIN Groups from CSV files:

Press button under “Options” to download/export PIN Group settings.

Inbound Routes

Inbound routes can be configured via Web GUI→Extension/Trunk→Inbound Routes.

Note: UCM6200 now supports 5000 Inbound route.

• Click on to add a new inbound route.

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• Click on "Blacklist" to configure blacklist for all inbound routes.

• Click on to edit the inbound route.

• Click on to delete the inbound route.

Inbound Rule Configurations

Table 60: 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.
Example: [12345-9] - Any digit from 1 to 9.
There are two pattern fields: Pattern and CallerID Pattern.

Pattern: Specifies the pattern that the dialed number must satisfy to use the
inbound route. Example: if the pattern is "_2xxx", the dialed number must be
a four-digit number that starts with 2 in order for this route to be used.

Pattern CallerID Pattern: Specifies the callerID pattern that callers must satisfy to
use the inbound route. Example: if the callerID pattern is "_1234", only
callers with a CID of 1234 can use this route.

Notes:
▪ Multiple patterns can be used. Each pattern should be entered in new
line.
▪ Users can add comments to the end of patterns to better organize and
keep track of complex rules by typing “/*” and “*/” before and after each
comment, respectively.
▪ Example:
Pattern CallerID Pattern
_X. 1000
_NNXXNXXXXX /* 10-digit long distance */ 1001
_818X. /* Any number with leading 818 */

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 After creating the inbound route, users can choose to enable and disable it. If
Disable This Route the route is disabled, it will not take effect anymore. However, the route
settings will remain in UCM. Users can enable it again when it is needed.
Allows the selected extension to use this function. If an extension is busy, and
Allowed to seamless transfer a mobile phone is bound to that extension, the mobile phone can pick up calls
to that extension.
If enabled, collect calls will be blocked.
Note 1: Collect calls are indicated by the header "P-Asserted-Service-Info:
service-code=Backward Collect Call, P-Asserted-Service-Info: service-
Block Collect Calls
code=Collect Call".
Note 2: There is also a global setting for this option in the SIP Settings ->
General Settings page.
Configure the Alert-Info, when UCM receives an INVITE request, the Alert-
Alert-Info
Info header field specifies an alternative ring tone to the UAS.
This option will allow users to choose the custom ring back tone to play when
Special ringing tone
caller reaches the route.
If enabled, the trunk name will be added to the caller id name as the displayed
Prepend Trunk Name
caller id name.
Manipulates Caller ID (CID) name and/or number within the call flow to help
Set Caller ID Info identify who is calling. When enabled two field will show allowing to
manipulate the CalleID Number and the Caller ID Name.
Configure the pattern-matching format to manipulate the numbers of incoming
callers or to set a fixed CallerID number for calls that go through this inbound
route.
• ${CALLERID(num)}: Default value which indicates the number of an
incoming caller (CID). The CID will not be modified.

• ${CALLERID(num):n}: Skips the first n characters of a CID number,

where n is a number.
CallerID Number • ${CALLERID(num):-n}: Takes the last n characters of a CID number,
where n is a number.
• ${CALLERID(num):s:n}: Takes n characters of a CID number starting
from s+1, where n is a number and s is a character position (e.g.
${CALLERID(num):2:7} takes 7 characters after the second character of
a CID number).

• n${CALLERID(num)}: Prepends n to a CID number, where n is a

number.
Default string is ${CALLERID(name)}，which means the name of an
CallerID Name incoming caller, it is a pattern-matching syntax format.

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 A${CALLERID(name)}B means Prepend a character 'A' and suffix a
character 'B' to ${CALLERID(name)}.
Not using pattern-matching syntax means setting fix name to incoming caller.
Gives uses the ability to configure inbound mode per individual route. When
enabled two field will show allowing to set the Inbound mode and the Inbound
Enable Route-Level Inbound mode Suffix.
Note: Global inbound mode must be enabled before users can configure
route-level inbound mode
Choose the inbound mode for this route.
Note: Toggling the global inbound mode will not affect routes that have Route-
Inbound Mode level Inbound Mode enabled. If all routes have the option enabled, toggling
the global inbound mode via BLF will trigger a voice prompt indicating that
none of the routes will be affected by the global inbound mode change.
Dial "Global Inbound Mode feature code + Inbound Mode Suffix" or a route's
assigned suffix to toggle the route's inbound mode.
Inbound Mode Suffix
The BLF subscribed to the inbound mode suffix can monitor the current
inbound mode.
Multiple mode allows user to switch between destinations of the inbound rule
by feature codes. Configure related feature codes as described in [Inbound
Inbound Multiple Mode
Route: Multiple Mode]. If this option is enabled, user can use feature code
to switch between different modes/destinations.
This option shows up only when "By DID" is selected. If enabled, the external
Dial Trunk users dialing in to the trunk via this inbound route can dial outbound call using
the UCM’s trunk.
This option shows up only when "By DID" is selected.

• Disable: Only the selected Extensions or Extension Groups are

allowed to use this rule, when enabled Filter on Source Caller ID.

• Internal: The lowest level required. All users are allowed to use this
rule, check this level might be risky for security purpose.
Privilege Level • Local: User with Local level, National or International level are
allowed to use this rule.

• National: Users with National or International Level are allowed to

use this rule.

• International: The highest level required. Only users with

international level are allowed to use this rule.

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 This option shows up only when "By DID" is selected. This controls the
destination that can be reached by the external caller via the inbound route.
The DID destination are:
• Extension
• Conference
• Call Queue
Allowed DID Destination • Ring Group
• Paging/Intercom Groups
• IVR
• Voicemail Groups
• Fax Extension
• Dial By Name
• All
Select the default destination for the inbound call.
• Extension
• Voicemail
• Conference Room
• Call Queue
• Ring Group
• Paging/Intercom
• Voicemail Group
• Fax
• DISA
Default Destination • IVR
• External Number
• By DID
When "By DID" is used, the UCM will look for the destination based on
the number dialed, which could be local extensions, conference, call
queue, ring group, paging/intercom group, IVR, 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.
• Dial By Name
• Callback
• Announcement

Specify the number of digits to strip from the beginning of the DID. This is
Strip
used when "By DID" is selected in "Default Destination".

Specify the digits to be prepended before the call is placed via the trunk.
Prepend
Those digits will be prepended after the dialing number is stripped.

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Time Condition
Start Time Select the start time "hour:minute" for the trunk to use the inbound rule.
End Time Select the end time "hour:minute" for the trunk to use the inbound rule.
Select either "By Week" or "By Month" for the time condition.
If "By Week" is selected, select the weekdays that the time condition will be
used.
If "By Month" is selected, select the months that the time condition will be
Frequency used. Next, select either "Week" or "Day".
(1) If "Week" is selected, choose the weekdays of the month that the time
condition will be used.
(2) If "Day" is selected, choose the specific days of the month that the time
condition will be used.
Select the destination for the inbound call under the defined time condition.
• Extension
• Voicemail
• Conference Room
• Call Queue
• Ring Group
• Paging/Intercom
• Voicemail Group
• Fax
• DISA
• IVR
Destination
• By DID
When "By DID" is used, the UCM will look for the destination based on
the number dialed, which could be local extensions, conference, call
queue, ring group, paging/intercom group, IVR, 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.
Configure the number of digits to be stripped in "Strip" option.
• Dial By Name
• External Number
• Callback
• Announcement

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:

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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.

Figure 123: 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 124: Inbound Route - Multiple Mode

When Multiple Mode is enabled for the inbound route, the user can configure a “Default Destination” and a “Mode
1” destination for all routes. By default, the call coming into the inbound routes will be routed to the default
destination.

SIP end devices that have registered on the 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).

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 Figure 125: Inbound Route - Multiple Mode Feature Codes

Inbound Route: Route-Level Mode

In the UCM6200, users can enable Route-Level Inbound Mode to switch between different destinations for each
individual inbound route. The inbound Route-Level mode can be enabled under Inbound Route settings.

Figure 126: Inbound Route - Route-Level Mode

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Global inbound mode must be enabled before configuring Route-Level Inbound Mode. Additionally, the Mode 1
must be configured as well.

When Route-Level Inbound Mode is enabled, the user can configure a “Default Destination” and a “Mode 1”
destination for each specific route. By default, the call coming into this specific inbound route will be routed to
the default destination.

Users can toggle the route’s inbound mode by dialing "Global Inbound Mode feature code + Inbound Mode
Suffix” and the current inbound route can be monitored by subscribing a BLF to the Inbound Mode Suffix.

For example, Inbound Default Mode feature code is set to *61 and the Inbound Mode suffix for route 1 is set to
1010. To switch the mode of route 1 to Default Mode, users can dial *611010.

Note: Toggling the global inbound mode will not affect routes that have Route-level Inbound Mode enabled. If
all routes have the option enabled, toggling the global inbound mode via BLF will trigger a voice prompt indicating
that none of the routes will be affected by the global inbound mode change.

Inbound Route: Inbound Mode BLF Monitoring

Users can assign MPKs and VPKs to monitor and toggle the current global inbound mode of the UCM.
To do this, please refer to the following steps:

1. Access the UCM web GUI and navigate to Extension/Trunk→Inbound Routes.

2. Click on the button and enable Inbound Multiple Mode.

3. Edit the subscribe number field to the desired BLF value.

Figure 127: Global Inbound Mode

4. Configure the BLF value on a phone’s MPK/VPK. As an example, a GXP2140 with the BLF configured
will show the Inbound Mode status on its screen once configured. The 777 BLF is lit green, indicating
that the current inbound mode is “Default Mode”.

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 Figure 128: Inbound Mode - Default Mode

5. Pressing the key will toggle the inbound mode to “Mode 1”, and the button’s color will change to red.

Figure 129: Inbound Mode - Mode 1

Inbound Route: Import/Export Inbound Route

Users can now import and export inbound routes to quickly set up inbound routing on a UCM or to back up an
existing configuration. An exported inbound route configuration can be directly imported without needing any
manual modifications.

Figure 130: Import/Export Inbound Route

The imported file should be on CSV format and using UTF-8 encoding, the imported file should contain below
columns, and each column should be separated by a comma (It is recommended to use Notepad++ for the
imported file creation):
• Disable This Route: Yes/No.
• Pattern: Always prefixed with _
• CallerID Pattern: Always prefixed with _

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 • Prepend Trunk Name: Yes/No.
• Prepend User Defined Name Enable: Yes/No.
• Prepend User Defined Name: A string.
• Alert-info: None, Ring 1, Ring 2… User should enter an Alert-info string following the values we have in
the Inbound route Alert-Info list.
• Allowed to seamless transfer: [Extension_number]
• Fax Detection: No, Yes.
• Fax Type: Extension, Fax to Email.
• Fax Destination: [Extension_number] or [Email address]
• Inbound Multiple Mode: Yes/No.
• Default Destination: By DID, Extension, Voicemail… User should enter a Default Destination string
following the values we have in the Inbound route Default Destination list.
• Destination: An Extension number, Ring Group Extension…
• Default Time Condition.
• Mode 1: By DID, Extension, Voicemail… User should enter a Default Destination string following the
values we have in the mode 1 Default Destination list.
• Mode 1 Destination: An Extension number, Ring Group Extension…
• Mode 1 Time Condition.

FAX Detection

The UCM can automatically detect fax signaling from FXO ports and forward these calls to specified destinations.

FAX with Two Media

Since UCM's system is now based on Asterisk 13, fax re-invites that negotiate with multiple codecs are now
supported. If a re-invite contains both T.38 and PCMU/PCMA codecs, T.38 will be prioritized.

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 or click on

button to remove all the numbers on the blacklist.

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• User can also export the inbound route blacklist by pressing on button.

Figure 131: Blacklist Configuration Parameters

• 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 132: Blacklist csv File

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Note:
Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature code
for "Blacklist Add' (default: *40) and "Blacklist Remove" (default: *41) from an extension. The feature code can
be configured under Web GUI→Call Features→Feature Codes.
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 CONFERENCE

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 61: Conference Room Configuration Parameters

Configure the conference number for the users to dial into the conference.
Extension
Note: The conference extension number can contain up to 64 characters.
When configured, the users who would like to join the conference call must enter
this password before accessing the conference room.

Notes:

Password • If "Public Mode" is enabled, the password is not required to join the conference
room thus this field is invalid.
• The password must contain at least 4 characters.
• Conference extension number can no longer be used as the conference
password if Strong Password is enabled.
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
Host Password the invited party) during the conference call.

Notes:
• If "Public Mode" is enabled, the password is not required to join the conference
room thus this field is invalid.

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 • Password must be at least 4 characters and different than conference
extension number.
If enabled, conference participant could press the * key to access the conference
Enable Caller Menu
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 will not be played. The default setting is "No".
Quiet Mode
Note:
"Quiet Mode" and "Announce Callers" cannot be enabled at the same time.

If there is only one participant in a conference room, a kick warning prompt will
Kick Warning play at the configured interval. If no input from the participant is received after the
Interval (minutes) prompt, he will be automatically kicked out of the conference. The valid range is
1-60 minutes.

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.
The default setting is "No".
Allow User Invite
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
the caller joins the conference. The default setting is "No".

Notes:
Announce Callers
• "Quiet Mode" and "Announce Callers" cannot be enabled at the same time.

• Conference participant names will be announced when joining/leaving the

conference even when the conference is on hold..

If enabled, no authentication will be required when joining the conference call. The
Public Mode
default setting is "Yes".

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 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 playlist to be played in conference call. Music On Hold
Music On Hold
playlist can be set up under Web GUI→PBX Settings→Music On Hold.

Custom Music On
Select a custom Music On Hold.
Hold

If enabled, the invitation from Web GUI for a conference room with password will
Skip Authentication
skip the authentication for the invited users. The default setting is "No".

Conference Settings contains the following options:

Table 62: 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.
Enable Neteq If enabled, conference audio quality may improve, but system
performance will be lowered.

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.

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 Figure 133: Conference

Conference Call Operations

Join a Conference Call

Users could dial the conference room extension to join the conference. If password is required, enter the
password to join the conference as a normal user, or enter the admin password to join the conference as
administrator.

Invite Other Parties to Join Conference

When using the 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 invite it to the conference.

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 Figure 134: 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 during the conference call. Please
make sure "Allow 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.

If 0 is entered, once the invited party picks up the invitation call, the system will ask the party to accept or reject
the invitation.
If 1 is entered, once the invited party picks up the call, they will automatically be brought into the conference.
----------------------------------------------------------------------------------------------------------------------------- ----------------------

Note:
Conference administrator can always invite other parties from the phone during the call by entering 0 or 1. To
join a conference room as administrator, enter the admin password when joining the conference. A conference
room can have multiple administrators.
---------------------------------------------------------------------------------------------------------------------------------------------------

During The Conference

During the conference call, users can manage the conference from Web GUI or IVR.

• Manage the conference call from Web GUI.

Log in 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.

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 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 63: 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.
• 1: List all users currently in the conference call.
• 2: Kick all non-Administrator participants from the conference call.
8 • 3: Mute/Unmute all non-Administrator participants from the conference call.
• 4: Record the conference call.
• 8: Exit the caller menu and return to the conference.

Conference User IVR Menu

1 Mute/unmute yourself.

4 Decrease the volume of the conference call.

5 Decrease your volume.

6 Increase the volume of the conference call.

7 Increase your volume.

8 Exit the caller menu and return to the conference.

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----------------------------------------------------------------------------------------------------------------------------- --------------------

Note:
When there is participant in the conference, the conference room configuration cannot be modified.
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Google Service Settings Support

UCM6200 now supports Google OAuth 2.0 authentication. This feature is used for supporting UCM6200
conference scheduling system. Once OAuth 2.0 is enabled, UCM6200 conference system can access Google
calendar to schedule or update conference.

Google Service Settings can be found under Web GUI→Call Features→Conference→Google Service
Settings→Google Service Settings.

Figure 135: Google Service Settings→OAuth2.0 Authentication

If you already have OAuth2.0 project set up on Google Developers web page, please use your existing login
credential for “OAuth2.0 Client ID” and “OAuth2.0 Client Secret” in the above figure for the UCM6200 to access
Google Service.

If you do not have OAuth2.0 project set up yet, please following the steps below to create new project and obtain
credentials:

1. Go to Google Developers page https://console.developers.google.com/start Create a New Project in Google

Developers page.

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 Figure 136: Google Service→New Project

2. Enable Calendar API from API Library.

3. Click “Credentials” on the left drop down menu to create new OAuth2.0 login credentials.

Figure 137: Google Service→Create New Credential

4. Use the newly created login credential to fill in “OAuth2.0 Client ID” and “OAuth2.0 Client Secret”.
5. Click “Get Authentication Code” to obtain authentication code from Google Service.

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 Figure 138: Google Service→OAuth2.0 Login

6. Now UCM6200 is connected with Google Service.

You can also configure the Status update, which automatically refresh your Google Calendar with the configured
time (m). Note: Zero means disable.

Conference Schedule

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 64: Conference Schedule Parameters

Schedule Options
Configure the topic of the scheduled conference. Letters, digits, _ and -
Conference Subject
are allowed.
Conference Room Select a conference room for this scheduled conference.
Conference Password Conference login password.
Host Password Host Password.

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 Set kick time before conference starts. When kick time is reached, a
warning prompt will be played for all attendees in the conference room.
After 5 minutes, this conference room will be cleared and locked for the
Kick Time(m) scheduled conference to begin.

Note: Kick Time cannot be less than 6 minutes in order to clear the
conference room.
If enabled, conference participants will not hear each other until the host
joins the conference.
Wait for Host
Note: If Quiet Mode is enabled, the voice prompt for this option will not
be played.
Description The description of scheduled conference.
Repeat Repeat interval of scheduled conference. By default, 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.
Duration of the conference meeting.
Meeting Duration
Note: The maximum allowed meeting duration that can be set is 8 hour(s).
Select this option to synchronize scheduled conference with Google
Calendar.
Enable Google Calendar
Note: Google Service Setting OAuth2.0 must be configured on the
UCM6200. Please refer to section [Google Service Settings Support].
Send email notification Sends Email notification to the extension.
Select the administrator of scheduled conference from selected
extensions.
Conference Administrator
Note: “Public Mode” must be disabled from Conference Room Options
tab.
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 in order to view
remote extensions here.
Add extensions that are not in the list (both local and remote list). If the
Special Extension user wishes to add the special extension, please make sure that the
UCM's outbound routes will allow calling that special extension.
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.

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 Configure the password to join as conference administrator.
Host Password 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
Enable Caller Menu
conference room menu by pressing the * key.
If this option is enabled, conference call will be recorded in .wav format.
Record Conference
The recorded file can be found from Conference page.
If this option is enabled, the notification tone or voice prompt for joining or
leaving the conference will not be played.
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 will not 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 will not be
played.
If this option is enabled, the user can:

• Press ‘0’ to invite others to join the conference with invited party’s
permission
• Press ‘1’ to invite without invited party’s permission
Allow User Invite
• Press ‘2’ to create a multi-conference room to another conference
room
• Press ‘3’ to drop all current multi-conference rooms

Note: Conference Administrator is always allowed to access this menu.

If this option is enabled, when a participant joins the conference room,
participant’s name will be announced to all members in the conference
Announce Callers room.
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
Play Hold Music one participant in the conference room, or the conference is not yet
started.
If this option is enabled, the invitation from Web GUI via a trunk with
Skip Authentication
password will not require authentication.

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 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.

After configuring the scheduled conference, it will be shown under Conference Schedule page as below:

Figure 139: Conference Schedule

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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.

Contact Group

Users can now quickly invite multiple participants at once to a conference via conference contact groups. Up to
5 contact groups can be created. The maximum allowed number of contacts per group is based on the UCM
model’s conference participant limit: 25 for 6202/6204, 32 for 6208, 64 for 6510.

Each contact group must have a password configured, which will be required when inviting the specified contact
group to a conference. Additionally, an audio file can be uploaded to each group to be used to announce the
contact group name such as “Sales” or “Marketing”. The default announcement for each group is “Conference
Contact Group 1”, “Conference Contact Group 2”, etc.

Figure 140: Contact Group Parameters

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Contact Group Configurations

• Click on "Create New Contact Group" to add a new Contact Group.

• Click on to edit the Contact Group.
• Click on to delete the Contact Group.

Table 65: Contact Group Parameters

Name Name associated to the contact group.

Password Password required to invite the specified contact group to a conference.

Audio file that can be uploaded to the group to announce the contact group
name such as “Sales” or “Marketing”.
Prompt
The default announcement for each group is “Conference Contact Group
1”, “Conference Contact Group 2”, etc.

Members Contacts that needs to be added in each group.

Type of the members to be added, it can be either Extensions or a self-

Type
defined number.

Conference Recordings

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 to download the recording or
click on to delete the recording. Users could also delete all recording files by clicking on “Delate All
Recording Files” or delete multiple recording files at once by clicking on “Delete Selected Recording Files” after
selecting the recording files.

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 Figure 141: Conference Recording

Conference Call Statistics

Conference reports will now be generated after every conference. These reports can be exported to a .CSV file
for offline viewing. The conference report page can be accessed by clicking on the Call Statistics button on the
main Conference page.

Figure 142: Conference Call Statistics

Figure 143: Conference Report on Web

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Figure 144: Conference Report on CSV

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 VIDEO CONFERENCE

With the UCM you can easily create, schedule, manage, and join video conference calls, from your desktop or
laptop computer. UCM Video conferencing uses WebRTC technology, so all the participants don’t have to
download and install any additional software or plugins. If upgrading from firmware that does not have this
functionality, system administrators must first toggle on the Enable WebRTC option for extensions that want to
use the UCM's WebRTC video conferencing feature. The video conference configurations can be accessed
under Web GUI→Call Features→Video Conference. Here, users can create and manage video conference
rooms and schedule video conferences.

Video 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 66: Video Conference room Configuration Parameters

Configure the conference number for the users to dial into the conference.
Extension
Note: Up to 64 characters.
When configured, the users who would like to join the conference call must
enter this password before accessing the conference room.

Note:
Password • Only digits are allowed.
• The password has to be at least 4 characters. All repetitive and
sequential digits (e.g., 0000, 1111, 1234 and 2345) or common digits
(e.g., 111222 and 321321) are not allowed.

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Conference Settings

Figure 145: Video Conference Basic settings

Basic Settings
This option should be enabled in order to activate the Video Conference
Video Conferencing
feature.
Configure the UDP port number for MCM. The standard UDP port for MCM
Bind UDP Port
is 5062.
Packet Loss Retransmission Configure to enable Packet Loss Retransmission.
If enabled, the Forward Error Correction (FEC) will be activated. The
FEC
default setting is "No".
If enabled, the AMI will send the corresponding event when a user starts
Enable Talk Detection
or stops talking.
The amount of time (ms) that sound exceeds what the DSP has
established as the baseline for silence before a user is considered to be
DSP Talking Threshold
talking. This value affects several operations and should not be changed
unless the impact on call quality is fully understood.
The amount of time(ms) that sound falls within what the DSP has
DSP Silence Threshold established as the baseline for silence before a user is considered be
silent. This value affects several operations and should not be changed

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 unless the impact on call quality is fully understood.

Conference Schedule

Conference Schedule can be found under UCM Web GUI → Call Features→Video Conference → Conference
Schedule. Users can create, edit, view and delete a Conference Schedule.

• Click on “Schedule New Conference” to add a new Conference Schedule.

• Click on the scheduled conference to edit or delete the event.
Table 67: Video Conference Schedule Parameters

Schedule Options
Configure the name of the scheduled conference. Letters, digits, _ and -
Conference Subject
are allowed.
Conference Room Select a conference room for this scheduled conference.
Configure conference room password. Please note that if “Public Mode” is
Conference Password
enabled, this option is automatically disabled.

Configure the time before the scheduled conference. When this time is
reached, a warning prompt will be played, and all attendees currently in
Kick Time(m) the scheduled conference room will be kicked after 5 mins. The conference
room will be locked until the scheduled conference begins. Default value
is 10 min.
Configure the beginning date and duration of scheduled conference.
Start Time Note: Please be mindful to avoid schedule conflicts for the same
conference room.

Host Set the admin of this scheduled conference from the following list of
members.

Repeat Choose when to repeat a scheduled conference.

Local Extension Select the extensions from the list to attend this scheduled conference.

Remote Extension The remote extension in the peer PBX connected to the local PBX via
LDAP sync.
Add extensions that are not in the list (both local and remote list). If the
Special Extension user wishes to add the special extension, please match the pattern on the
outbound route.
Description Set a description of scheduled conference.
Time Zone Configure the conference Time Zone.
Once created, the Web GUI will display scheduled conference in Conference Schedule.

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Please see figure below:

Figure 146: Video Conference Schedule

Once the conference room is scheduled, at the kick time, all users will be removed from conference room and
no extension is allowed to join the conference room anymore. At the scheduled conference time, UCM will send
INVITE to the extensions that have been selected for conference.

--------------------------------------------------------------------------------------------------------------------------------------------

Notes:
• Video conferencing can be resource-intensive and may cause performance issues with the UCM when used.
• To ensure the best experience, please use Google Chrome (v67 or higher) or Mozilla Firefox (v60).
----------------------------------------------------------------------------------------------------------------------------- ---------------

Wave WebRTC Video Calling & Conferencing

Web audio and video calls and conferencing can now be achieved through the UCM’s new WebRTC page. To
get started with this new feature, please make sure to:

1. Navigate to Value-Added Features → WebRTC and enable WebRTC support.

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 Figure 147: Enabling WebRTC Feature

2. Enable the WebRTC on the extensions that would use this feature under Extension / Trunk →
Extensions.

Figure 148: Enabling WebRTC on Extensions

The UCM offers the possibility to log in to an extension via Grandstream Wave Portal using user portal password
in addition to SIP registration password, where it offers a sleek interface to host conferences, receive email
reminders for scheduled conferences, manage contacts, initiate calls, call transfer, chat functionality and more.

Access the page by adding “/gswave” after the UCM’s server address and port. (e.g.
https://my.ucm.com:8089/gswave).

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 Figure 149: Grandstream Wave Interface

Note: Starting with 1.0.19.27, the registration limit was increased to 300 for UCM62xx. The limit is the same
regardless of whether the user is making voice calls or video calls

For more details about the WebRTC feature, please refer to the following guide:
http://www.grandstream.com/sites/default/files/Resources/wave_webrtc_guide.pdf

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 IPVIDEOTALK MEETINGS

UCM extensions can now dial into IPVT (IPVideoTalk) meetings by creating a peer trunk to an IPVT server.
However, users must make sure that the IPVT server they are peering to also has a peer trunk to their UCM
configured. This setting can be found in Admin Center→SIP Trunk Configuration.

Figure 150: IPVT SIP Trunk page

Next, users must create a peer trunk on the UCM to the IPVT server. Enter one of the following addresses based
on the desired connection protocol:
• TCP: pro.ipvideotalk.com:20000
• TLS: pro.ipvideotalk.com:20001

Figure 151 - Peer Trunk to IPVT

Make sure that the Transport field is either “TCP” or “TLS”. Save and apply changes to create the trunk.

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Next, edit the newly created trunk and click on the Advanced Settings tab. Make sure the IPVT Mode option is
checked. Otherwise, you may experience audio issues when dialing into IPVT.

Figure 152 - IPVT Mode

Finally, create an outbound route for this trunk. This route will be used to dial IPVT meeting IDs. Due to IPVT
meeting IDs having a random assortment of numbers, it is recommended to use a unique code to precede the
meeting ID so that UCM can direct calls to the IPVT trunk without fail (e.g., *99). In the below image, “x.” would
be the meeting ID.

Figure 153 - IPVT Outbound Pattern

However, if a unique code is used, users must also configure the Strip field to remove the unique code from the
meeting ID before the call is sent to IPVT.

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 Figure 154 - IPVT Outbound Strip

In this example, the Strip field has “3” configured to remove the example unique code *99 from the dialed number
before the call is routed out to the IPVT server. Once this outbound route has been created, users can now use
a UCM extension to dial IPVT meeting rooms.

Note: An IPVT account can have only 1 SIP trunk peered to it.

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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.

Figure 155: Create New IVR

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 Table 68: 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, the caller is allowed to dial extensions other than the ones
Dial other Extensions
explicitly defined.
This option shows up only when "By DID" is selected. This controls the
destination that can be reached by the external caller via the inbound
route. The DID destination are:
• Extension
• Conference
• Call Queue
DID Destination
• Ring Group
• Paging/Intercom Groups
• Voicemail Groups
• Fax Extension
• Dial By Name
• All
If enabled, all callers to the IVR is allowed to use trunk. The permission
Dial Trunk must be configured 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 into the IVR, the
Permission
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.
If enabled, the UCM will replace the caller display name with the IVR name
Replace Caller ID 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
Alert-Info
and alternative ring tone to the UAS.
Select an audio file to play as the welcome prompt for the IVR. Click on
"Prompt" to add additional audio file under Web GUI→PBX
Welcome Prompt
Settings→Voice Prompt→Custom Prompt.
Note: Up to 5 welcome prompts can be used.
Digit Timeout Configure the timeout between digit entries. After the user enters a digit,

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 the user needs to enter the next digit within the timeout. If no digit is
detected within the timeout, the UCM6200 will consider the entries
complete. The default timeout is 3 seconds.
After playing the prompts in the IVR, the UCM6200 will wait for the DTMF
entry within the timeout (in seconds). If no DTMF entry is detected within
Response Timeout
the timeout, a timeout prompt will be played. The default setting is 10
seconds.
Response Timeout Prompt Select the prompt message to be played when timeout occurs.
Select the prompt message to be played when an invalid extension is
Invalid Input Prompt
pressed.
Configure the number of times to repeat the prompt if no DTMF input is
Response Timeout Prompt
detected. When the loop ends, it will go to the timeout destination if
Repeats
configured, or hang up. The default setting is 3.
Configure the number of times to repeat the prompt if the DTMF input is
Invalid Input Prompt Repeats invalid. When the loop ends, it will go to the invalid destination if
configured, or hang up. 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
Language 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.
Key Pressing Events
Select the event for each key pressing for 0-9, *, Timeout and Invalid. The
Key Press Event: event options are:
Press 0 • Extension
Press 1 • Voicemail
Press 2 • Conference Rooms
Press 3 • Voicemail Group
Press 4 • IVR
Press 5 • Ring Group
Press 6 • Queues
Press 7 • Page Group
Press 8 • Fax
Press 9 • Custom Prompt
Press * • Hangup
Timeout • DISA
Invalid • Dial By Name
• External Number

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 • Callback
• Announcement

Figure 156: Key Pressing Events

IVR Black/Whitelist

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.

Note: up to 500 extensions are allowed on the black/white list.

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
allowed to be reached via IVR.

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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 157: 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 158: 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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 VOICE PROMPT

The UCM6200 supports multiple languages in Web GUI as well as system voice prompt. The following languages
are currently supported in system voice prompt:

English (United States), Arabic, Chinese, Dutch, English (United Kingdom), French, German, Greek, Hebrew,
Italian, Polish, Portuguese, Russian, Spanish, Catalan, Swedish, Czech 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. Additionally, UCM6200 allows to customize specific prompt instead of full
language package, and it provides ability to upload greeting files for extensions.

Language Settings

Download and Install Voice Prompt Package

To download and install voice prompt package in different languages from UCM6200 Web GUI, click on "Add
Voice Prompt Package" button.

Figure 159: 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 160: 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 161: New Voice Prompt Language Added

A new language option will be displayed after successfully installed. Users then could select it to apply in the
UCM6200 system voice prompt or delete it from the UCM6200.

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Upload Language Package

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 162: Upload Voice Prompts Package

The package file should follow below requirements:

▪ Each file uploaded must be under 50MB.
▪ Package structure:
[Package]
├ [voice prompt dir]
│ ├ [... dir]
│ └ [... files]
└ info.txt (containing the language name for display, in UTF8)
▪ Language dir name format:
▪ Custom dir name format: language_xxx;
For example: If there is a Chinese custom directory named zh_xxx, the custom voice prompt in zh_xxx
would be used first, then the Chinese voice prompt zh, then use the default language prompt (en); If
not named the format as above, then the custom prompt will be used first, then use the default language
prompt (en).

For more details, please refer to:

http://www.grandstream.com/sites/default/files/Resources/ucm_voiceprompt_customization_guide.zip

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Custom Prompt

Record New Custom Prompt

In the UCM6200 Web GUI→PBX Settings→Voice Prompt→Custom Prompt page, click on "Record New IVR
Prompt" and follow the steps below to record new IVR prompt.

Figure 163: Record New IVR 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.

Upload Custom Prompt

If the user has a pre-recorded IVR prompt file, click on "Upload IVR 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.
• Filename should not exceed 100 characters.

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 Figure 164: Upload IVR Prompt

Click on “choose file to upload” to select audio file from local PC and to start uploading. Once uploaded, the file
will appear in the IVR Prompt web page.

Download All Custom Prompt

On the UCM6200, 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 165: Download All Custom Prompt

Note: The downloaded file will have a .tar extension.

Username Prompt Customization

There are two ways to customize/set new username prompt:

Upload Username Prompt File from Web GUI

1. First, users should have a pre-recorded file respecting the following format:

• PCM encoded / 16 bits / 8000Hz mono.

• In “.GSM” or “.WAV” format.
• File size under 5M.

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 • Filename must be set as the extension number with 18 characters max. For example, the recorded
file name 1000.wav will be used for extension 1000.

2. Go under web GUI PBX Settings → Voice Prompt → Username Prompt and click on

button.

3. Select the recorded file to upload it and press Save and Apply Settings.

4. To delete username prompts, either click on the button to delete a single file or select multiple files
and click on Delete button to delete multiple prompts at once.

Record Username via Voicemail Menu

The second option to record username is using voicemail menu, please follow below steps:

• Dial *98 to access the voicemail

• After entering the desired extension and voicemail password, dial “0” to enter the recordings menu and

then “3” to record a name.

Another option is that each user can record their own name by following below steps:

• The user dials *97 to access his/her voicemail

• After entering the voicemail password, the user can press “0” to enter the recordings menu and then “3”

to record his name.

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 VOICEMAIL

Configure Voicemail

If the voicemail is enabled for UCM6200 extensions, the configurations of the voicemail can be globally set up
and managed under Web GUI→Call Features→Voicemail.

Figure 166: Voicemail Settings

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 Table 69: Voicemail Settings

Configure the maximum number of seconds for the voicemail greeting. The
Max Greeting
default setting is 60 seconds.
If enabled, the caller can press 0 to exit the voicemail application and connect
Dial ‘0’ For Operator
to the configured operator’s extension.
Select the operator extension, which will be dialed when users press 0 to exit
Operator Extension
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:
• 1 minute
Max Message Time • 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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 If enabled, external callers routed by DID and reaching VM will be prompted by
the UCM with 2 options:

• Press 1 to leave a message.

To leave a message for the extension reached by DID.
Voicemail Remote
Access • Press 2 to access voicemail management system.
This will allow caller to access any extension VM after entering
extension number and its VM password.

Note: This option applies to inbound call routed by DID only.

The default setting is “Disabled”.

Forward Voicemail to Enables the forwarding of voicemail to remote extensions on peered SIP trunks.
Peered UCMs The default setting is “Disabled”.

Configures the default voicemail password that will be used when an extension
Voicemail Password is reset.

Note: Resetting an extension will reset Voicemail Password, Send Voicemail to Email, and Keep Voicemail after
Emailing values to default. Previous custom voicemail prompts and messages will be deleted.

Access Voicemail

If the voicemail is enabled for UCM6200 extensions, the users can dial the voicemail access number (by default
*97) to access their extension’s voicemail. The users will be prompted to enter the voicemail password and then
can enter digits from the phone keypad to navigate in the IVR menu for different options.

Otherwise, the user can dial the voicemail access code (by default *98) followed by the extension number and
password in order to access to that specific extension’s voicemail.

Table 70: 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

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 * - 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
2 - Record your busy message 1 - Accept this recording
2 - Listen to it
3 - Re-record your message
3 - Record your name 1 - Accept this recording
2 - Listen to it
3 - Re-record your message
4 - Record temporary greeting 1 - Accept this recording
2 - Listen to it
3 - Re-record your message
5 - Change your password
* - Return to the main menu

Leaving Voicemail

If an extension has voicemail enabled under basic settings “Extension/Trunk → Extensions → Basic Settings”
and after a ring timeout or user not available, the caller will be automatically redirected to the voicemail in order
to leave a message on which case they can press # in order to submit the message.

In case if the caller is calling from an internal extension, they will be directly forwarded to the extension’s
voicemail box.

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But if the caller is calling from outside the system and the incoming call is routed by DID to the destination
extension, then the caller will be prompted with the choice to either press1 to access voicemail management or
press 2 to leave a message for the called extension. This feature could be useful for remote voicemail
administration.

Voicemail Email Settings

The UCM6200 can be configured to send the voicemail as attachment to Email. Click on "Voicemail Email
Settings" button to configure the Email attributes and content.

Table 71: Voicemail Email Settings

Attach Recordings to If enabled, voicemails will be sent to user's Email address. The default setting
E-Mail is "Yes".

If enabled, voicemail will be stored in the UCM6200 after the email is sent. The
Keep Recordings
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 167: Voicemail Email Settings

Click on "Load Default Settings" button to view the default template as an example.

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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.

Figure 168: Voicemail Group

Table 72: Voicemail Group Settings

Enter the Voicemail Group Extension. The voicemail messages left to this
Voicemail Group Extension
extension will be forwarded to all the voicemail group members.
Configure the Name to identify the voicemail group. Letters, digits, _ and -
Name
are allowed.
Voicemail Password The Voicemail password for the user to check Voicemail messages.
Email Address The Email address of current user.
Select available extensions from the left list and add them to the right list.
The extensions need to have voicemail enabled to be listed in available
Member
mailboxes list.
Note: Members selected cannot exceed 30 extensions.

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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 169: Ring Group

• Click on to add ring group.

• Click on to edit the ring group. The following table shows the ring group configuration parameters.

• Click on to delete the ring group.

Table 73: Ring Group Parameters

Configure ring group name to identify the ring group. Letters, digits, _ and –
Ring Group Name
are allowed.
Extension Configure the ring group extension.
Select available users from the left side to the ring group member list on the
Members
right side. Click on to arrange the order.
Select available remote users from the left side to the ring group member list
LDAP Phonebook on the right side. Click on to arrange the order. Note: LDAP Sync must
be enabled first.
Select the ring strategy. The default setting is “Ring in order”.

• Ring simultaneously.
Ring Strategy Ring all the members at the same time when there is incoming call to the
ring group extension. If any of the member answers the call, it will stop
ringing.

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 • Ring in order.
Ring the members with the order configured in ring group list. If the first
member does not answer the call, it will stop ringing the first member and
start ringing the second member.

Select the “Music On Hold” Class of this Ring Group, “Music On Hold” can be
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 the users to upload the customized voice
prompts.
Custom Prompt
Note: Users can also refer to the page PBX Settings→Voice
Prompt→Custom Prompt, where they could record new prompt or upload
prompt files.

Configure the number of seconds to ring each member. If set to 0, it will keep
ringing. The default setting is 30 seconds.
Ring Timeout for Each
Member 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.
This allows the UCM to work with endpoint-configured call forwarding settings
to redirect calls to ring group. For example, if a member wants to receive calls
to the ring group on his mobile phone, he would have to set his endpoint’s call
forwarding settings to his mobile number. By default, it is disabled.

However, this feature has the following limitations:

• This feature will work only when call forwarding is configured on

Endpoint Call endpoints, not on the UCM.

Forwarding Support • If the forwarded call goes through an analog trunk, and polarity reversal
is disabled, the other ring group members will no longer receive the call
after it is forwarded.
• If the forwarded call goes through a VoIP trunk, and the outbound route
for it is PIN-protected and requires authentication, the other ring group
members will no longer receive the call after it is forwarded.
• If the forwarded call hits voicemail, the other ring group members will no
longer receive the call.

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 If enabled, the UCM will replace the caller display name with the Ring Group
Replace Display Name name the caller know whether the call is incoming from a direct extension or a
Ring Group.

If enabled calls to busy agents will be skipped and sent to the following
Skip Busy Agent
available ones. Default is enabled.
If enabled, users could select extension, voicemail, ring group, IVR, call queue,
voicemail group as the destination if the call to the ring group has no answer.
Enable Destination Note: Voicemail Password and Email address (limited by 128 characters) are
required if voicemail is selected as the destination. Voicemail system will
mention if a voicemail is from a ring group.
The call would be routed to this destination if no one in this ring group answers
Default Destination
the call.

Figure 170: Ring Group Configuration

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Remote Extension in Ring Group

Remote extensions from the peer trunk of a remote UCM6200 can be included in the ring group alongside local
extensions. An example of Ring Group with peer extensions is presented in the following:

1. Create SIP trunks from UCM A to UCM B and vice-versa on the Extension/Trunk->VoIP Trunks page.
Additionally, please create the appropriate outbound and inbound routes for these SIP trunks.

2. Click edit button in the menu , and check if Sync LDAP Enable is toggled on.

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Figure 171: Sync LDAP Server option

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3. In the scenario that the LDAP is not synced automatically, users can manually sync LDAP phonebooks by
clicking on the LDAP sync button that's highlighted in the following image.

Figure 172: 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 the LDAP phonebook has synced correctly, the remote UCM's extensions should appear in the
Create/Edit Ring Group->LDAP Phonebook list.

Figure 173: Ring Group Remote Extension

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 RESTRICT CALLS

Restrict calls is a feature that can be used to restrict calls between internal extensions besides those in the
Allowed List.

This section describes the configuration of this feature in the Call Features->Restrict Calls page.

Figure 174: Restrict Calls

Configure Restrict Calls

• Click on "Add" to add a rule for restrict calls.

• Click on to edit the rule of restrict calls.

• Click on to delete the rule of restrict calls.

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 Table 74: Restrict Calls Configuration Parameters

Name Configure Restrict call’s name.

Restrict Calls between When enabled, members of the group cannot dial other extension, only the
extensions numbers in the Allowed List.
Configure the members that will not be able to call any extensions besides those
Members
in the Allowed List.
Allowed list Select the extensions that the Members list can be able to call.

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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 "Add" to add paging/intercom group.

• Click on to edit the paging/intercom group.

• Click on to delete the paging/intercom group.

• Click on "Paging/Intercom Group Settings" to edit Alert-Info Header. This header will be included in the SIP
INVITE message sent to the callee in paging/intercom call.

Configure Multicast Paging

Figure 175: Multicast Paging

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 Table 75: Multicast Paging Configuration Parameters

Name Configure paging/intercom group name.

Type Select “Multicast Paging”.

Extension Configure the paging/intercom group extension.

The allowed multicast IP address range is 224.0.1.0 - 238.255.255.255.
Multicast IP Address
Note: This field appears only when “Type” is set to “Multicast Paging”.
Specify the maximum call duration in seconds. The default value 0 means no
Maximum Call Duration
limit.
This option is to configure a custom to play to paging callees. Users can now
Custom Prompt directly upload custom prompts from this page. They do not need to be
redirected anymore.
Multicast IP Address Configure the multicast IP address that endpoints will listen to.
Configure the port for multicast paging.
Port
Note: This field appears only when “Type” is set to “Multicast Paging”.

Configure 2-way Intercom

Figure 176: 2-way Intercom

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 Table 76: 2-way Intercom Configuration Parameters

Name Configure paging/intercom group name.

Type Select "2-way Intercom".

Extension Configure the paging/intercom group extension.

If enabled, the UCM will replace the caller display name with Paging/Intercom
Replace Display Name
name.
Specify the maximum call duration in seconds. The default value 0 means no
Maximum Call Duration
limit.
This option is to set a custom prompt for a paging/intercom group to announce
Custom Prompt to caller. Users can now directly upload custom prompts from this page. They
do not need to be redirected anymore.
Select available users from the left side to the paging/intercom group member
Members
list on the right.
Paging/Intercom Select which extensions are allowed to use the paging/intercom feature for this
Whitelist paging group.

Configure 1-way Paging

Figure 177: 1-way Paging

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 Table 77: 1-way Paging Configuration Parameters

Name Configure paging/intercom group name.

Type Select "1-way Paging".

Extension Configure the paging/intercom group extension.

If enabled, the UCM will replace the caller display name with Paging/Intercom
Replace Display Name
name.
Specify the maximum call duration in seconds. The default value 0 means no
Maximum Call Duration
limit.
This option is to set a custom prompt for a paging/intercom group to announce
Custom Prompt to caller. Users can now directly upload custom prompts from this page. They
do not need to be redirected anymore..
Select available users from the left side to the paging/intercom group member
Members
list on the right.
Paging/Intercom Select which extensions are allowed to use the paging/intercom feature for this
Whitelist paging group.

Configure Announcement Paging

Figure 178: Announcement Paging

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 Table 78: Announcement Paging Configuration Parameters

Enable Enable/Disable Announcement Paging.

Name Configure paging/intercom group name.

Type Select "Announcement Paging”

This option is to set a custom prompt for a paging/intercom group to announce

Custom Prompt to caller. Users can now directly upload custom prompts from this page. They
do not need to be redirected anymore.
Repeat If enabled, the announcement page will be repeated for the selected weekdays.
Date Configure Announcement Paging Date.
Include Holidays If enabled, Announcement Paging will run during holidays.
Time Configure Announcement Paging Time.
Configure Announcement Paging transmission method.
Transmission Method Unicast: Depending on members selection
Multicast: Depending on Multicast IP address and Port
Select available users from the left side to the paging/intercom group member
Members
list on the right.

Configure Private Intercom

Private Intercom is a new paging type that is meant to be used with Grandstream GSC3510.
http://www.grandstream.com/products/facility-management/intercoms-paging/product/gsc3510

In a private intercom:

• The initiator can be heard by all parties

• The initiator can hear only one of the intercom members, which is determined by whose audio is initially
detected. Audio from other members cannot be heard until the first responder is done talking.

• Intercom members can hear only the initiator’s audio and not other intercom members

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 Figure 179: Private Intercom

Table 79: Private Intercom Configuration Parameters

Name Configure paging/intercom group name.

Type Select "Private Intercom".

Extension Configure the paging/intercom group extension.

If enabled, the UCM will replace the caller display name with Paging/Intercom
Replace Display Name
name.

Specify the maximum call duration in seconds. The default value 0 means no
Maximum Call Duration
limit.

This option is to set a custom prompt for a paging/intercom group to announce

to caller. Click on ‘Prompt’, it will direct the users to upload the customized voice
Custom Prompt prompts.
Note: Users can also refer to the page PBX Settings→Voice Prompt→Custom
Prompt, where they could record new prompt or upload prompt files.

Select available users from the left side to the paging/intercom group member
Members
list on the right.

Paging/Intercom Select which extensions are allowed to use the paging/intercom feature for this
Whitelist paging group.

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Paging/Intercom Group Settings

Figure 180: 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.

Configure a Scheduled Paging/Intercom

Users can schedule paging/intercom calls by using the Schedule Paging/Intercom page. To schedule, click the
Add button on the new page and configure the caller, the group to use, and the time to call out.

Figure 181: Schedule Paging/Intercom page

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 Table 80: Schedule Paging / Intercom Settings

Caller Configure the caller ID for the paging / intercom group.

Paging/Intercom Group Select the paging / intercom group from the list of the available groups.

Start Time Configure the start time of the scheduled paging / intercom call.

Select the type for the scheduled paging / intercom call. The available types are:
Type
Single time or Daily, Weekly basis. Default is “Single”.
Include Holidays If enabled Paging/Intercom will run during holidays.
Action Status Display the action status of the scheduled paging / intercom call.

Figure 182: Creating a scheduled paging/intercom call

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 CALL QUEUE

The UCM provides call center functionality through its Call Queue feature. Static and dynamic agents can be
assigned to queue groups to handle varying loads of call traffic. This section describes the configuration of call
queues in the Call Features->Call Queue page.

Configure Call Queue

Call queue settings can be accessed via Web GUI→Call Features→Call Queue.

Figure 183: 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 81: 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
Ring all available Agents simultaneously until one answer.

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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 playlist for the call queue.
Music On Hold Note: Music On Hold playlists can be managed from Web GUI→PBX
Settings→Music On Hold.
Configure the maximum number of calls that can be queued at once. This number
does not include calls that have been connected with agents. It only includes calls
Max Queue Length not connected yet. The default setting is 0, which means unlimited. When the
maximum value is reached, the caller will be treated with busy tone followed by
the next calling rule after attempting to enter the queue.
Configure the number of seconds an agent must wait to receive a new call after
Wrapup Time completing a previous call. If set to 0, an agent can receive a new call as soon
as a previous call is completed. Default is 10 seconds.
Retry Time Configure the number of seconds to wait before ringing the next agent.
Configure the number of seconds to ring an agent before proceeding to ring the
Ring Time
next agent. Default 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.
If enabled, call queue members can use recording feature code to pause or
Enable Record
restore current queue's recording when Auto Record is enabled for the queue.
Feature Code
Default code for Start/Stop Call Recording is *3.
If enabled, users can upload an audio file that will be played as an initial tone
Welcome Prompt
when dialing the queue number.
Configure the maximum amount of seconds callers can wait before being
disconnected from the queue. Default setting is 60 seconds. If left blank, there
Max Wait Time
will be no limit to the amount of time a caller waits in queue.
Note: It is recommended to configure "Wait Time" longer than the "Wrapup Time".

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 Configure the destination that callers will be redirected to after the configured Max
Destination
Wait Time has passed. Default is "Hang up".
Reset Agent Call If enabled, the agent call completion count will be reset based on the frequency
Counter - Enable set in the Repeat and Date/Time fields.
Repeat Specifies the frequency at which the Agent call counter will be reset.
Destination Prompt Configure the frequency to play the Destination Prompt to callers waiting in
Cycle queue.
The custom prompt to play to callers waiting in queue once the Destination
Prompt Cycle period has been reached. Callers can press 1 to be redirected to
Custom Prompt
the destination configured in Destination.

Select the failover destination to redirect callers to after pressing 1 during 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
Enables quick static agent login/logout via GXP21XX softekys. Supports
Enable Agent Login
GXP21XX endpoints on firmware 1.0.9.18 and higher.
Configure whether or not callers will be disconnected from the queue based on
agent availability.

• 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 there are no agents logged in
or if all agents are paused 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 the queue if it has no agents logged in or if all agents

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 in it are paused or unavailable.
Configure the destination to redirect callers to based on the Leave When
Empty and Dial in Empty Queue values.
• Play Sound.
• Extension.
• Voicemail.
Failover Destination
• Queues.
• Ring Group.
• Voicemail Group.
• IVR.
• External Number.
If enabled, the UCM will report to the agent the amount of time a caller waited
Report Hold Time
before being connected. Default is disabled.
If enabled, the UCM will replace the caller display name with the Call Queue name
Replace Display Name
so that the caller knows the call is incoming from a Call Queue.
Enable feature codes option for call queue. For example, *83 is used for “Agent
Enable Feature Codes Pause”.
Note: Callers can no longer use feature codes in established callbacks.
Dynamic Login If enabled, the configured PIN number is required for dynamic agent to log in. The
Password default setting is disabled.
Configure the call destination for the call to be routed to if no agent in this call
Alert-Info
queue answers the call.
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.

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 Figure 184: Static Agents limit

The following table lists the maximum number of static agents for each UCM model:

Table 82: Static Agent Limitation

UCM Model Max Static Agents in Call Queue

UCM6202 22

UCM6204 23

UCM6208 34

UCM6510 75

Click on "Global Queue Settings" to configure Agent Login Extension Postfix and Agent Logout Extension
Postfix. Once configured, users could log in the call queue as dynamic agent.

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 Figure 185: Agent Login Settings

For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout Extension
Postfix is **, users could dial 6500* to login to the call queue as dynamic agent and dial 6500** to logout from
the call queue. Dynamic agent does not need to be listed as static agent and can log in/log out at any time.

• Call queue feature code "Agent Pause" and "Agent Unpause" can be configured under Web 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 to
download the recording file in .wav format; click on to delete the recording file. To delete multiple
recording files by one click, select several recording files to be deleted and click on “Delete Selected
Recording Files” or click on “Delete All Recording Files” to delete all recording files.

Call Center Settings and Enhancements

UCM supports additional call center features such as virtual call queue and position announcement, allowing
callers to know their current position in the queue and providing them the option to either stay in the queue or
request for callback once an agent is available.

To configure these, click on the button for a queue group and click on the Advanced Settings tab. The
following options are available:

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 Table 83: Call Center Parameters

Enable Virtual Queue Enable virtual queue to activate call center features.
Configure the amount of seconds a caller must wait before the virtual queue
Virtual Queue Period
prompt is played to them. Default is 20 seconds.
Timeout Mode: After the virtual queue period passes, the caller will enter the
virtual call queue and be presented with a menu to choose an option, the choices
are summarized below :
• Press * to set current number as callback number.
• Press 0 to set a callback number different than current caller number.
• Press # to keep waiting on the call queue.
Virtual Queue Mode Note: These options are customizable from the Global Queue Settings page.

DTMF Mode: In this mode, the callers can activate the virtual queue by pressing
2, then they will be presented with the menu to choose an option as below:
• Press * to set current number as callback number.
• Press 0 to set a callback number different than current caller number.
• Press # to keep waiting on the call queue.
Virtual Queue
System will add this prefix to dialed numbers when calling back users.
Outbound Prefix
Enable Virtual Queue If enabled, agents will have a set amount of time to answer a virtual queue
Timeout callback.
Write Timeout Configure the virtual queue callback timeout period in seconds.
If enabled, callers' positions in queue will be announced based on the frequency
Enable Position set in Position Announcement Interval.
Announcement Note: Queue position will now be announced to the caller upon entering the
queue.
Position
Announcement Configure the frequency of the caller position announcement.
Interval
Enable Hold time Enable the announcement of the estimated hold time to the caller periodically.
Announcement Note: Hold time will not be announced if less than one minute.
Select the extension to act as chairman of the queue (monitoring).
Queue Chairman
Note: One queue can have up to 3 chairmen.
Enables quick static agent login/logout via GXP21xx softkeys. Supports
GXP21XX endpoint on firmware 1.0.9.18 and higher.
Notes:
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” to display a softkey on the phone to login/logout.

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 ✓ When this option is enabled, dynamic agent login will be no longer
supported.
✓ In case of concurrent registrations, changing agent status on one phone
(login/logout) will be reflected on all phones.
Toggles the way UCM handles and distributes call traffic. Please see the section
Autofill
below for more details.

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
wait in the queue even if there is more than one available member ready to take calls until the head caller is
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 the mentioned call center features, UCM offers detailed call queue statistics, allowing system
administrators to make informed decisions regarding call traffic distribution and handling.

Figure 186: Call Queue Statistics

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Select the time interval along with the queue(s) and agent(s) to get detailed statistics.

User can download statistics on CSV format by clicking on the download button , also the statistics

can be cleared using “Clear Statistics” button .

The statistics can be automatically sent to a specific email address on a preconfigured Period, this can be done

by clicking on “Automatic Download Settings” button , and user will be directed to

below page where he can configure the download period (Day/Week/Month) and the Email where the statistics
will be sent (Email settings should be configured correctly):

Figure 187 : Automatic Download Settings - Queue Statistics

Significantly more information is now available UCM’s queue statistics page. In addition to the information
presented in previous firmware, users can now view a call log that displays calls to all agents and queues, a
dynamic agent login/logout record, and a pause log. Statistics reports for these new pages can be obtained by
pressing the Download button in the top left corner of the Call Queue Statistics page. The reports are in .CSV
format and will be packaged into a single tar.gz file upon download.

Agent Details

Agent Details is a call log that shows every call to each individual agent from all queues. The following
information is available:

• Time – the date and time the call was received.

• Agent – the agent that was rung for the call.
• Queue – the queue that the call went to.
• Caller ID Number – the CID of the caller
• Abandoned – indicates whether the call was picked up or not by that specific agent. If the call rang
several agents simultaneously, and this specific agent did not pick up the call, the call will be considered
abandoned even if a different agent in the same queue picked it up.
• Wait Time – the amount of time that the call was waiting in queue after dialing in.
• Talk Time – the duration of the call after it was picked up by agent.

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 Figure 188: Agent details

Login Record

Login Record is a report that shows the timestamps of dynamic agent logins and logouts and calculates the
amount of time the dynamic agents were logged in. Dynamic agents are extensions that log in and out either via
agent login/logout codes (configured in Global Queue Settings page) or by using the GXP21xx call queue
softkey. A new record will be created only when an agent logs out. The following information is available:

• Agent – the extension that logged in and out.

• Queue – the queue that the extension logged in and out of.
• Login Time – the time that the extension logged into the queue.
• Logout Time – the time that the extension logged out of the queue.
• Login Duration – the total length of time that the extension was logged in.

Figure 189: Login Record

Pause Log

Pause Log is a report that shows the times of agent pauses and unpauses and calculates the amount of time
that agents are paused. If an agent is part of several queues, an entry will be created for each queue. An entry
will only be created after an agent unpauses. The following information is available:

• Agent – the extension that paused and unpaused

• Queue – the queue that the agent is in.
• Pause Time – the time that the agent paused.
• Resume Time – the time that the agent unpaused.
• Pause Duration – the total length of time the agent was paused for.

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 Figure 190: Pause Log

Global Queue Settings

As explained before, under this section users can configure the feature codes for Dynamic agent login and
logout, and also can now customize the keys for virtual queue options like shown below.

Figure 191: Global Queue Settings

Table 84: Global Queue Settings

Dynamic Agent Login Settings

Configure the code to dial after the queue extension to log into the queue (i.e.
Agent Login Code
queue extension + suffix).
Suffix
If no suffix is configured, dynamic agents will not be able to log in
Configure the code to dial after the queue extension to log out of the queue (i.e.
Agent Logout Code
queue extension + suffix).
Suffix
If no suffix is configured, dynamic agents will not be able to log out.

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Virtual Queue Callback Key Settings
Call Back Current
Press the feature key configured to set your current number as callback number.
Number
Custom Callback
Press the feature key configured to set a custom callback number.
Number
Continue Waiting Press the feature key configured to continue waiting.

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Switchboard

Switchboard is a tool that allows system administrators and users to monitor and manage queues. This can be
accessed from the Call Features->Call Queue page.

Figure 192: Switchboard Summary

Clicking on one of the queues shown above will bring up the following window:

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 Figure 193: Call Queue Switchboard

The table below gives a brief description for the main menus:

Table 85: Switchboard Parameters

This menu shows the current waiting calls along with the caller id and the option
Waiting
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.
Displays the list of agents in the queue and the extension status (idle, ringing, in
use or unavailable) along with some basic call statistics and agent’s mode (static
or dynamic).
Agents
Note: the dashboard will show the number of calls (answered and abandoned) of
each agent. For dynamic agents, it will count the number of calls starting from the
last login time.

There are three different privilege levels for Call Queue management from the switchboard: Super Admin, Queue
Chairman, and Queue Agent.

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• Super Admin - Default admin of the UCM. Call queue privileges include being able to view and edit all
queue agents, monitor and execute actions for incoming and ongoing calls for each extension in
Switchboard, and generate Call Queue reports to track performance.

• Queue Chairman - User appointed by Super Admin to monitor and manage an assigned queue extension
via Switchboard. The Queue Chairman can log into the UCM user portal with his extension number and
assigned user password. To access the Switchboard, click on “Value-added Features” in the side menu and
click on “Call Queue”. In the image below, User 1012 is the Queue Chairman appointed to manage Queue
Extension 6500 and can see all the agents of the queue in the Switchboard.
Note: Super Admin can assign 3 Chairmen

• Queue Agent - User appointed by Super Admin to be a member of a queue extension. A queue agent can
log into the UCM user portal with his extension number and assigned user password. To access the
Switchboard, click on “Value-added Features” in the side menu and click on “Call Queue”. However, a queue
agent can view and manage only his own calls and statistics, but not other agents’ in the queue extension.
In the image below, User 1000 is a queue agent and can see only his own information in the Switchboard.

Figure 194: Queue Agent

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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 to create a new pickup group.

• Click on to edit the pickup group.

• Click on to delete the pickup group.

Select extensions from the list on the left side to the right side.

Figure 195: Edit Pickup Group

Configure Pickup Feature Code

When picking up the call for the pickup group member, the user only needs to dial the pickup feature code. It is
not necessary to add the extension number after the pickup feature code. The pickup feature code is configurable
under Web 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 196: 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 playlist and upload music files. The "default" Music On Hold playlist already has
5 audio files defined for users to use.

Figure 197: Music On Hold Default Playlist

• Click on "Create New MOH Class" to add a new Music On Hold playlist.

• Click on to configure the MOH class sort method to be "Name" or "Random" for the sound files.

• Click on next to the selected Music On Hold playlist to delete this Music On Hold playlist.

• Click on to start uploading. Users can upload:

➢ Single files with 8KHz Mono Music file, or

➢ Music on hold files in a compressed package with .tar, .tar.gz and .tgz as the suffix. The file name can
only be letters, digits or special characters -_
➢ the size for the uploaded file should be less than 30M, the compressed file will be applied to the entire
MoH.
• Users could also download all the music on hold files from UCM. In the Music On Hold page, click on

and the file will be downloaded to your local PC.

• Click on next to the sound file to disable it from the selected Music On Hold Playlist.
• Click on next to the sound file to enable it from the selected Music On Hold Playlist.

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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 UCM can play selected MOH files to extensions by initiating calls to them from the Manage Music on Hold
page

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 198: 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 199: Information Prompt

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 3. Click .

4. In the prompted window, select the extension to playback and click .

Figure 200: 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.

----------------------------------------------------------------------------------------------------------------------------- --------------------

Notes:
Once the MOH file is deleted, there are two ways to recover the music files.
• Users could download the MOH file from this link:
http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-moh-opsound-wav-2.03.tar.gz
After downloading and unzip the pack, users could then upload the music files to UCM.
• Factory reset could also recover the MOH file on the UCM.
------------------------------------------------------------------------------------------------------------------------------------------ --------

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 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 201: Fax Settings

Table 86: FAX/T.38 Settings

Enable Error Configure to enable Error Correction Mode (ECM) for the Fax.
Correction Mode The default setting is "Yes".
Configure the maximum transfer rate during the Fax rate negotiation.
Maximum Transfer
The possible values are 2400, 4800, 7200, 9600, 12000 and 14400.
Rate
The default setting is 14400.

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 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.

• Only
Max Concurrent Under this mode, the UCM6200 allows only single user to send fax at a time.
Sending Fax • More
Under this mode, the UCM6200 supports multiple concurrent fax sending by
the users.

By default, this option is set to “only”.

Configure the maximum length of Fax Queue from 6 to 10.
Fax Queue Length
The default setting is 6.
User Information in Fax If enabled this this will give users the option to send a special header in SIP fax
Header messages.
Fax Header Information Adds fax header into the fax file.
Configure the Email address to send the received Fax to if user's Email address
cannot be found.

Default Email Address 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
Enables the fax resend option which allow the UCM to keep attempting to send
faxes up to a specified amount of times. Additionally, if a fax still fails to send, a
Enable Fax Resend
Resend button will appear in the File Send Progress list in Value-Added
Features→Fax Sending to allow manual resending.
Max Resend Attempts Configures the maximum attempts number to resend the fax.

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Receiving Fax

Example 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 202: 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 is set to "None".
• Once selected, analog related settings for this extension will show up in "Analog Settings" section.

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 Figure 203: Configure Extension for Fax Machine: FXS Extension

Figure 204: 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 205: 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.

Example 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.

Figure 206: 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 207: 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 208: 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 209: Fax Sending in Web GUI

After that you can see the ongoing sending operation on the progress bar.

Figure 210: Fax Send Progress

Note: Only A3, A4, and B4 paper sizes are supported for the Fax Sending.

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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 supports SIP presence, allowing extensions to advertise their current availability for calls. This presence
can be monitored by other users.

Presence is different from BLF in that SIP presence is a status that users can manually set themselves.

Users can change presence status from the webUI:

Admin Portal: Navigate to Extension/Trunk->Extensions->Edit Extension->Features and click on the

Presence Status dropdown list.

User Portal: Navigate to Basic Information->Extensions->Features and click on the Presence Status
dropdown list.

Figure 211: SIP Presence Configuration

Select which status to set from the presence status selection drop list.
The following options are available:

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 Table 87: 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.
Please enter the presence status for this mode on the Web GUI. Up to 64
Custom Presence Status
characters.
Unavailable The contact is unreachable for the moment, please try to contact later.

Users can also set presence status by dialing the feature code from their phones (*48 by default). The prompt
will ask users what presence status they want to set.

The feature code can be enabled and customized from the Web GUI→Call Features→Feature Codes.

Figure 212: SIP Presence Feature Code

When a user does change his/her SIP presence status by making a call using presence feature code, the UCM
will create a corresponding CDR entry showing the call as Action type = PRSENCE_STATUS.

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 Figure 213: Presence Status CDR

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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 214: 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.

The following table shows the Follow Me configuration parameters:

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 Table 88: 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
Confirm When to 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.
When enabled, the call will be routed to the default destination if no one in the
Enable Destination
Follow Me extensions answers the call.
Configure the destination if no one in the Follow Me extensions answers the call.
The available options are:
• Extension
• Voicemail
Default Destination • Queues
• Ring Group
• Voicemail Group
• IVR
• External Number
The added numbers are listed here. Click on to arrange the order. Click
Follow Me Numbers
on to delete the number. Click on to add new numbers.
Add a new Follow Me number which could be a ‘Local Extension’ or ‘External
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 89: 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 specified destinations by pressing only 1-4
digits. 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 create a speed dial extension and assign a destination to it.

Supported destinations are Extension, Conference Room, Video Conference Room, Voicemail, Voicemail
Group, IVR, Ring Group, Queue, Paging/Intercom Group, Fax, DISA, Dial by Name, Announcement, and
External Number.

Figure 215: Speed Dial Destinations

Figure 216: List of Speed Dial

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 DISA

In many situations, the user will find the need to access his own IP PBX resources, but he is not physically near
one of his extensions. However, he does have access to his own cell phone. In this case, we can use what is
commonly known as DISA (Direct Inward System Access). Under this scenario, the user will be able to call from
the outside, whether it is using his cell phone, pay phone, regular PSTN, etc. After calling into 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 to add a new DISA.

• Click on to edit the DISA configuration.

• Click on to delete the DISA.

Figure 217: Create New DISA

The following table details the parameters to set and configure DISA feature on UCM6200 PBX.

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 Table 90: 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.
Configure the maximum amount of time permitted between digits when the user is
Digit Timeout
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 is *0) to disconnect the call or hang up directly. A new dial tone
will be heard shortly for the user to make a new call. The default setting is "No".
Replace Display
If enabled, the UCM will replace the caller display name with the DISA name.
Name

Once successfully created, users can configure the inbound route destination as "DISA" or IVR key event as
"DISA". When dialing into DISA, users will be prompted with password first. After entering the correct password,
a second dial tone will be heard for the users to dial out.

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 EMERGENCY

UCM supports configuration and management of numbers to be called in emergency situation, thus bypassing
the regular outbound call routing process and allowing users in critical situation to dial out for emergency help
with the possibility to have redundant trunks as point of exit in case one of the lines is down.

UCM6xxx series are also now in full compliance with Kari’s Law and Ray Baum’s Act, for more information,
please refer to the following links:

https://www.fcc.gov/mlts-911-requirements

http://www.grandstream.com/sites/default/files/Resources/UCM_Emergency_Calls_Guide.pdf

In addition, Emergency calls can be automatically recorded by toggling on the new Auto Record and recordings
can be viewed in the new Emergency Recordings tab on the same page. Additionally, users can have these
recordings be sent to the configured email address(es).

Email alerts are also supported after enabling the notification for the event under “Maintenance → System
Events”

To configure emergency numbers, users need to follow below steps:

1. Navigate on the web GUI under “Call Features → Emergency Calls”

2. Click on to add a new emergency number.

3. Configure the required fields “Name, Emergency Number and Trunk(s) to be used to reach the number”.

4. Save and apply the configuration.

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 Figure 218: Emergency Number Configuration

The table below gives more description of the configuration Parameters when creating emergency numbers.

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 Table 91: Emergency Numbers Parameters

Configure the name of the emergency call.

Name
For example, "emergency911","emergency211" and etc.

Emergency Number Config the emergency service number. For example,"911","211" and etc.

Select the emergency level of the number. Level "3" means the most
Emergency Level
urgent.

If this option is not enabled, when the lines of trunks which the coming
emergency call routes by are completely occupied, the line-grabbing
Disable Hunt on Busy function will automatically cut off a line from all busy lines so that the
coming emergency call can seize it for dialing out. This option is not
enabled by default.

Configures a custom prompt that will be played to the notified extensions

Custom Prompt when an emergency call is made. The file can be uploaded from the page
"Custom Prompt". Click "Upload Audio File" to add a custom prompt.

Select the trunks for the emergency call. Up to five trunks can be
Use Trunks
selected.

Members Notified Select the members who will be notified when an emergency call occurs.
Specify the number of digits that will be Stripped from the beginning of the
dialed number before the call is placed via the selected trunk.
Strip
Note: Users can strip the same amount of numbers as the emergency
number length itself.

Specify the digits to be Prepended before the call is placed via the trunk.
Prepend
Those digits will be prepended after the dialing number is stripped.

Auto Record When enabled, emergency call will be automatically recorded.

Send Recording File When enabled recording files will be sent to the configured email address.

Configure the email addresses that will receive emergency call

Email Address
notifications.

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 Figure 219: 911 Emergency Sample

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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 92: Callback Configuration Parameters

Name Configure a name to identify the Callback. (Enter at least two characters)
Configure the pattern of the callers allowed to use this callback. The caller who
places the inbound call needs to have the CallerID match this pattern so that the
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
The caller can then enter the desired number to dial out via UCM6200 trunk.

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 BLF AND EVENT LIST

BLF

The UCM6200 supports BLF monitoring for extensions, ring group, call queue, conference room and parking lot.
For example, on the user's phone, configure the parking lot number 701 as the BLF monitored number. When
there is a parked call on 701, the LED for this BLF key will light up in red, meaning a call is parked against this
parking lot. Pressing this BLF key can pick up the call from this parking lot.

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Note:
On the Grandstream GXP series phones, the MPK supports "Call Park" mode, which can be used to park the
call by configuring the MPK number as call park feature code (e.g., 700). MPK "Call Park" mode can also be
used to monitor and pickup parked call if the MPK number is configured as parking lot (e.g., 701).
-------------------------------------------------------------------------------------------------------------------------------------------- -------

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.

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 Figure 220: Create New Event List

Table 93: 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 -.
Select the available extensions/Extension Groups listed on the local UCM6200
Local Extensions
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.

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 Manually enter the remote extensions in the peer/register trunk to be monitored
Special Extensions
in the event list. Valid format: 5000,5001,9000

Event List BLF can be used to monitor remote LDAP extensions on VoIP trunks. SIP endpoints will need to
support event list BBLF to monitor remote extensions.

When an event list is created on the UCM and remote extensions are added to the list, the UCM will send out
SIP SUBSCRIBE to the remote UCM to obtain the remote extension status. When the SIP end points register
and subscribe to the local UCM 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.

---------------------------------------------------------------------------------------------------------------------------------------------------

Notes:
• 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 UCMs.

• (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.

Figure 221: Create Dial by Name Group

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 Figure 222: Configure Extension First Name and Last Name

1. Name
Enter a Name to identify the Dial By Name group.

2. Extension
Configure the direct dial extension for the Dial By Name group.

3. Custom Prompt

This option sets a custom prompt for directory to announce to a caller. The file can be uploaded from the
page "Custom Prompt". Click "Upload Audio File" to add additional record.

4. Available Extensions/Selected Extensions

Select available extensions from the left side to the right side as the directory for the Dial By Name group.
Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group.
The extensions here must have a valid first name and last name configured under Web
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.

5. Prompt Wait Time

Configure “Prompt Wait Time” for Dial By Name feature. During Dial By Name call, the caller will need to
input the first letters of First/Last name before this wait time is reached. Otherwise, timeout will occur, and
the call might hang up. The timeout range is between 3 and 60 seconds.

6. Query Type
Specify the query type. This defines how the caller will need to enter to search the directory.
By First Name: enter the first 3 digits of the first name to search the directory.
By Last Name: enter the first 3 digits of the last name to search the directory.
By Full Name: enter the first 3 digits of the first name or last name to search the directory.

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7. Select Type

Specify the select type on the searching result. The IVR will confirm the name/number for the party the caller
would like to reach before dialing out.

By Order: After the caller enters the digits, the IVR will announce the first matching party's name and number.
The caller can confirm and dial out if it is the destination party, or press * to listen to the next matching result
if it is not the desired party to call.

By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can press
number 1 to 8 to select and call or press 9 for results in next page.

The Dial by Name group can be used as the destination for inbound route and key pressing event for IVR. The
group name defined here will show up in the destination list when configuring IVR and inbound route. If Dial by
Name is set as a key pressing event for IVR, users can dial star (*) to exit from Dial by Name, and re-enter the
IVR to start a new event. The following example shows how to use this option.

Figure 223: Dial By Name Group In IVR Key Pressing Events

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 Figure 224: Dial By Name Group In Inbound Rule

Username 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 username prompt for an extension:

Upload Username Prompt File from Web GUI

1. First, Users should have a pre-recorded file respecting the following format:

• PCM encoded / 16 bits / 8000Hz mono.

• In .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 → Username Prompt and click on

button.

3. Select the recorded file to upload it and press Save and Apply Settings.

• Click on to record again the username prompt.

• Click on to play recorded username prompt.

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 • Select username prompts and press to delete specific file or select multiple files for deletion using

the button .

Record Username via Voicemail Menu

Users can also record their username via the voicemail menu.

• Dial *97 or *98 to access the voicemail menu. If *98 is dialed, enter the desired extension and
voicemail password.

• Upon entering the extension's voicemail system, dial 0 to enter the recordings menu.

• Press 3 to start recording the username.

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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 1000 is calling 1001. 1001 is ringing.

Figure 225: Status→PBX Status→Active Calls - Ringing

The following figure shows the call between 1002 and 1003 is established.

Figure 226: Status→PBX Status→Active Calls – Call Established

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The black color of the active call means the connection of call time is less than half an hour. It means this call is
normal.

Figure 227: Call Connection less than half hour

The yellow color of the active call means the connection of call time is greater than half an hour but less than
one hour. It means this call is a bit long.

Figure 228: Call Connection between half an hour and one hour

The red color of the active call means the connection of call time is more than one hour. It means this call could
be abnormal.

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 Figure 229: Call Connection more than one hour

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 230: 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”.

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 • Listen
In “Listen” mode, the extension monitoring the call can hear both parties in the active call but the audio
of the user on this extension will not be heard by either party in the monitored active call.

• Whisper
In “Whisper” mode, the extension monitoring the call can hear both parties in the active call. The user
on this extension can only talk to the selected monitored extension and he/she will not be heard by the
other party in the active call. This can be usually used to supervise calls.

• Barge
In “Barge” mode, the extension monitoring the call can talk to both parties in the active call. The call will
be established similar to three-way conference.

4. Enable or disable “Require Confirmation” option. If enabled, the confirmation of the invited monitor’s
extension is required before the active call can be monitored. This option can be used to avoid adding
participant who has auto-answer configured, or call forwarded to voicemail.

5. Click on “Add”. An INVITE will be sent to the monitor’s extension. The monitor can answer the call and start
monitoring. If “Require Confirmation” is enabled, the user will be asked to confirm to monitor the call.

Another way to monitor active calls is to dial the corresponding feature codes from an extension. Please refer to
[Table 94: UCM6200 Feature Codes] and [Call Recording] section for instructions.

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 CALL FEATURE CODES

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 94: 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
complete the attended transfer. In case of the called party does
not answer, users could press *0 to cancel the call and retrieve
Attended Transfer
the first call leg.
• Options:
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.
• Default code: *44 (Disabled by default).
• Seamless Transfer allows user to perform blind transfer using
UCM feature code without having music on hold presented
during the transfer process, it minimizes the interruption during
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
perform the seamless transfer.

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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.
Start/Stop Call Recording • 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.

Set the maximum interval (ms) between digits for feature code
Feature Code Digits Timeout
activation
DND/Call Forward

Do Not Disturb (DND) Activate • Default code: *77

Do Not Disturb (DND) Deactivate • Default code: *78

• Default Code: *90

Call Forward Busy Activate • Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.

Call Forward Busy Deactivate • Default Code: *91

• Default Code: *92

Call Forward No Answer Activate • Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.

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Call Forward No Answer
• Default Code: *93
Deactivate

• Default Code: *72

Call Forward Unconditional
• Enter the code and follow the voice prompt. Or enter the code
Activate
followed by the extension to forward the call.

Call Forward Unconditional

• Default Code: *73
Deactivate
• If enabled, this option will allow specific extensions to dial the
Remote Call Forward Enable remote call forwarding feature codes to set call forwarding for
any extension.

• Default Code: *65

Remote Call Forward Busy • Enter the code and follow the voice prompt to set the
Enable remote extension number where you want to enable Call
Forward Busy and the target destination.

• Default Code: *651

Remote Call Forward Busy • Enter the code and follow the voice prompt to set the
Disable remote extension number where you want to disable the
Call Forward Busy.

• Default Code: *66

Remote Call Forward No Answer • Enter the code and follow the voice prompt to set the
Enable remote extension number where you want to enable Call
Forward No Answer and the target destination.

• Default Code: *661

Remote Call Forward No Answer • Enter the code and follow the voice prompt to set the
Disable remote extension number where you want to disable the
Call Forward No Answer.

• Default Code: *67

Remote Call Forward • Enter the code and follow the voice prompt to set the
Unconditional Enable remote extension number where you want to enable Call
Forward Unconditional and the target destination.

• Default Code: *671

Remote Call Forward • Enter the code and follow the voice prompt to set the
Unconditional Disable remote extension number where you want to disable the
Call Forward Unconditional.

Only the Extensions selected in this whitelist can configure call

Remote Call Forward Whitelist
forwarding for any extension via feature codes.

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Feature Codes
• Default Code: *98
• Enter *98 and follow the voice prompt. Alternatively, dial *98
Voicemail Access Code
followed by the extension and pound (#) to immediately access
the entered extension's voicemail box.

• Default Code: *97

My Voicemail
• Press *97 to access the voicemail box.

• Default Code: *83

Agent Pause
• Pause the agent in all call queues.

• Default Code: *84

Agent Unpause
• Unpause the agent in all call queues.

• Default Code: *81

Paging Prefix • To page an extension, enter the code followed by the extension
number.

• Default Code: *80

Intercom Prefix • To intercom an extension, enter the code followed by the
extension number.

• Default Code: *40

Blacklist Add • To add a number to blacklist for inbound route, dial *40 and
follow the voice prompt to enter the number.

• Default Code: *41

Blacklist Remove • To remove a number from current blacklist for inbound route,
dial *41 and follow the voice prompt to remove the number.

• Default Code: **
Call Pickup on Ringing • To pick up a call for any extension xxxx, enter the code followed
by the extension number xxxx.

• Default Code: *45 (Disabled by default).

• If “Pickup In-call” feature is enabled, only the extensions added
Pickup In-call
in “Allowed to seamless transfer” in the extension’s Seamless
Transfer Privilege Control List” can pick up the call.

• Default Code: *8
• This code is for the pickup group, which can be assigned for
each extension on the extension configuration page.
Pickup Extension
• 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.

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 • 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.

• Default Code: *88

• If you have the permission to call mobile phone number, use
Direct Dial Mobile Phone Prefix
this prefix plus the extension number can dial the mobile phone
number of this extension directly.
• Default Code: *11
Call Completion Request • This code is for the user who wants to use Call Completion to
complete a call.
• Default Code: *12
Call Completion Cancel • This code is for the user who wants to cancel Call Completion
request.
Check this box to enable spy feature codes.
Enable Spy
Disabled by default.
• Default Code: *54 (“Enable Spy” needs to be checked)
• This is the feature code to listen in on a call to monitor
Listen Spy
performance. Monitor’s line will be muted, and neither party will
hear from the monitor’s extension.
• Default Code: *55 (“Enable Spy” needs to be checked)
• This is the feature code to speak to one side of the call (for
Whisper Spy
example, whisper to employees to help them handle a call).
Only one side will be able to hear from the monitor’s extension.
• Default Code: *56 (“Enable Spy” needs to be checked)
Barge Spy • This is the feature code to join in on the call to assist both
parties.
• Default Code: *36
Wakeup Service • Dial this code to access UCM wakeup service, you can add,
update, activate or deactivate wakeup service.
• Default Code: *35
PMS Wakeup Service • Dial this code to access UCM PMS wakeup service, you can
add, update, activate or deactivate PMS wakeup service.

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 • Default Code: *23
• Use this code with maid code to update PMS room status.
Update PMS Room Status
Choose the status to set after hearing the prompt, for example:
for maid 001 dial *23001 and then 1 after hearing the prompt.
• Dial this code to set the presence status of the extension.
• Possible options are:
1:"unavailable"
2:"available"
Presence Status
3:"away"
4:"chat
5:"dnd
6:"userdef"

• Default Code: *85

Dynamic Agent Logout
• Use this code to logout the dynamic agent from all queues.

Call voicemail group access code to access group voicemail. If

Voicemail Group Access Code
password is required, enter password followed by pound “#” key.

The UCM6200 also allows user to one click enable / disable specific feature code as shown below:

Figure 231: Enable/Disable Feature codes

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Parking Lot

User can create parking lots and their related slots under Web GUI→ Call Features→ Parking Lot. In the
Parking Lot page, users can create lots of their own. This allows different groups within an organization to have
their own parking lots instead of sharing one large parking lot with others. While creating a new parking lot, users
can assign it a range that they think is appropriate for the group that will use the parking lot.

Figure 232: Parking Lot

User can create a new Parking lot by clicking on button “Create New Parking Lot” :

Figure 233: New Parking Lot

Table 95 : Parking Lot

• Default Extension: 700

Parking Lot Extension • During an active call, initiate blind transfer and then enter this
code to park the call.

Parking Lot Name • Set a name to the parking lot

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 • Default Extension: 701-720
Parked Slots • 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.

• Default setting is 300 seconds and the maximum limit is 99.999

seconds.
Parking Timeout (s) • 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 Playlists Select the Music on Hold Class.

Configures a callback failover destination when the extension that

is called back is busy. The call will be routed to the destination
Failover Destination
number and this reduces the chance of dropping parked calls.
Note: This field cannot exceed 32 characters.
If enabled, all registered endpoints of the extension will ring when
callback occurs. Otherwise, only the original endpoint will be called
Ring All Callback on Timeout back.
Note: This option will not be available if Forward to Destination on
Timeout is enabled.
Forward to destination on If enabled, the call will be routed to the configured destination upon
timeout timeout. Otherwise, the call will be routed back to the original caller.
This option appears once Forward to Destination on Timeout is
Timeout Destination enabled. Upon park timeout, the call will be routed to the configured
destination.

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 before parking the call.

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• 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.

Monitor Call Park CID Name Information (GXP21xx Phones Only)

Users can see the CID name information of parked calls.

VPK/MPKs must be configured as “Monitored Call Park” with
the desired parking lot extension. The display will alternate
between displaying the parking lot extension and the call’s
CID name. There is no need to configure anything on the
UCM.

Note: This feature requires Grandstream GXP21xx new

firmware support. Please check GXP21xx firmware release
Figure 234: Monitored call park CID name
information for availability.

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 “Start/Stop Call Recording" is configured and enabled.
2. After establishing the call, enter the "Start/Stop Call Recording" feature code (by default it is *3) followed by
# or SEND to start recording.
3. To stop the recording, enter the "Start/Stop Call Recording" feature code (by default it is *3) followed by # or
SEND again. The recording will also be stopped upon hanging up the call.

4. The recording file can be retrieved under Web GUI→CDR. Click on to show and play the recording or

click on to download the recording file.

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 Figure 235: Download Recording File from CDR Page

The above recorded call's recording files are also listed under the UCM6200 Web GUI→CDR→Recording Files.
Note: Starting with firmware 1.0.20.17, Music on Hold will be also included in the recording.

Figure 236: Download Recording File from Recording Files Page

Enable Spy

If “Enable Spy” option is enabled, feature codes for Listen Spy, Whisper Spy and Barge Spy are available for
users to dial from any extension to perform the corresponding actions.
Assume a call is on-going between extension A and extension B, user could dial the feature code from extension
C to listen on their call (*54 by default), whisper to one side (*55 by default), or barge into the call (*56 by default).
Then the user will be asked to enter the number to call, which should be either side of the active call, extension
A or B in this example.
-------------------------------------------------------------------------------------------------------------------------------------------- -------

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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Shared Call Appearance (SCA)

Shared Call Appearance (SCA) functionality has been added to the UCM. With SCA, users can assign multiple
devices to one extension, configure endpoints to monitor that extension, make actions on behalf of that extension
such as viewing call status and placing and receiving calls, and even barging into existing calls. To configure the

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SCA functionality, please follow the steps below:

1. Users can enable SCA by navigating to the Extensions page, editing the desired extension, and enabling
the option SCA.
Note: With SCA enabled, the Concurrent Registrations field can only have a value of 1.

Figure 237: Enabling SCA option under Extension’s Settings

2. After enabling the option, navigate to Call Features→SCA. The newly enabled SCA extension will be listed.
Click the “+” button under the Options column to add a number that will share the main extension’s call
appearance, which will be called private numbers.

Figure 238: SCA Number Configuration

3. Configure the private number as desired.

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 Figure 239: SCA Private Number Configuration

4. Once the private number has been created, users must now register a device to it. To properly register a
device to the private number, use the configured private number as the SIP User ID. Auth ID and Password
will be the same as the main extension’s. Once registration is complete, SCA is now configured.

Figure 240: SCA Options

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SCA has various options to change its behavior:

• Allow Call Retrieve from Another Location – Allows users to retrieve held calls using any device
associated with the SCA extension.

• Alert All Appearances for Group Paging Calls – Alerts all devices associated with the SCA extension.

• Multiple Call Arrangement – Allows all devices associated with the SCA extension to make different
calls at the same time.

• Allow Bridging between Locations – Allows devices associated with the SCA extension to barge into
existing calls of the same SCA group.

• Bridge Warning Tone – Notification sound that will play when a party barges into the call. Three options
are available:

o None – No notification sound will play

o Barge-In Only – The notification sound will play once when a party barges in.

o Barge-In and Repeat – The notification sound will play when a party barges in and will play
again after every 30 seconds.

5. Next, configure the VPK or MPK to Shared for both the main extension and the private number. SCA is
now configured for both endpoint devices.

The following table describe the SCA Number configuration setting:

Table 96: Add SCA Private Number

Private Number Configures the private number for the SCA.

Related Shared Line Display the related shared line.

Enable This Number Enable/Disable the private number.

Allow Origination from This Enabling this option will allow calling from this private number. By
Number default, it is enabled.

Allow Termination to This Enabling this option will allows calls to this private number. By
Number default, it is enabled.

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The following table describes the options available when editing the SCA number:

Table 97: Editing the SCA Number

While SCA is enabled, this number will be the same as the

Shared Line Number
extension number.
Allow Call Retrieve from Another Allows remote call retrieval. Must be enabled in public hold. By
Location default, it is enabled.
Allows all SCA group members to ring when the SCA shared
Alert All Appearances for Group
number is paged. If disabled, only the SCA shared number will ring
Paging Calls
when paged. By default, it is disabled.

Multiple Call Arrangement Allows simultaneous calls in an SCA group. By default, it is disabled.

Allow Bridging between Allows location bridging for SCA group. Must be enabled when
Locations using the Barge-In feature. By default, it is disabled.
Configures the notification in the bridge when another party join.
• None: No notification sound.
• Barge-In only: Notification sound will play when another
Bridge Warning Tone party join.
• Barge-In and Repeat: Notification sound will play when
another party joins and repeat every 30 seconds.
By default, it is set to “Barge-In Only”.

Announcement

The Announcement feature (not to be confused with Announcement Paging and Announcement Center) is a
feature that allows users to set an unskippable audio file to play to callers before routing them to a configured
destination. Announcements can be configured as a destination in the [Inbound Routes] or in [IVR].

To configure the Announcement, users need to follow below steps:

1. Navigate on the web GUI under “Call Features → Announcement”

2. Click on to add a new Announcement.

3. Configure the required fields Name, Prompt, Default Destination to be used for the announcement.

4. Save and apply the configuration.

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 Figure 241: Announcement settings

The table below gives more description of the configuration parameters when creating Announcement.

Table 98: Announcement Parameters

Configure the name of the Announcement.

Name Note: Please use letters, digits, _ or – only and no more than 64
characters.

Audio file that will be played before ringing the configured default
destination.

Note: Sound file must be PCM encoded, 16 bits at 8000Hz mono in

mp3/wav format or raw ulaw/alaw/gsm file with
Prompt
.mp3/.wav/.ulaw/.alaw/.gsm suffix. The file size must be less than 5MB.
If uploading a compressed file, the file must have .tar/.tgz/.tar.gz suffix. the
file name must contain only letters, numbers or special characters -_. The
file size must be less than 30MB.
Filename cannot exceed 100 characters.

Select the destination where to send the call after playing the
announcement.
The available default destinations are:
• Extension
Default Destination • Conference Rooms
• Video Conference
• Voicemail
• Voicemail Group
• IVR

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 • Ring Group
• Queues
• Fax
• DISA
• Dial By Name
• External Number
• Hang-up

Created Announcements will be listed as shown below:

Figure 242: Announcement

• Press to edit the announcement.

• Press to delete the announcement.

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 PBX SETTINGS

This section describes internal options that have not been mentioned in previous sections yet. The settings in
this section can be applied globally to the 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.

PBX Settings/General Settings

Table 99: 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.
Configure the global CallerID Name used for all outbound calls. If configured,
Global Outbound CID
all outbound calls will have the CallerID Name set to this name. If not, the
Name
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".
Reroutes failed outbound calls that receive a 486 response through the failover
Enable 486 to Failover
trunk to retry the call. If disabled, calls that receive a 486 response will be
Trunk
terminated.
Enter a name to identify the UCM. The name will be displayed on UCM web
Device Name
interface.
Configure the International prefix. Default is 00.
If empty, international call prefix can be empty or +.
This parameter helps the UCM to identify the international call prefix for the
country (00, 011, 810…) to avoid any conflict when using blacklist for specific
countries.
International Call Prefix
When outbound blacklist is enabled, UCM will apply the following rule
"International Call Prefix+Country Code+Destination number". When making
outbound call, UCM will check "International Call Prefix” then “Country Code",
if the combination exists in the blacklist, then the call will fail, otherwise, the call
will be authorized.

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 For example: If blacklist is selecting countries with prefix code 001, dialing a
number like 00123456789 will be blocked even if the number is not part of
blacklisted countries if “International Call Prefix” is not set. While in same
example, if “International Call Prefix” is set to “00”, UCM will allow the dialed
number.
Extension Preferences
If enabled, strong password will be enforced for the password created on the
UCM6200. The default setting is enabled.
Strong Password Rules:
1. Password for voicemail, voicemail group, outbound route, DISA, call queue
and conference require non-repetitive and non-sequential digits, with a
minimum length of 4 digits. Repetitive digits pattern (such as 0000, 1111,
1234, 2345, and etc.), or common digits pattern (such as 111222, 321321
Enforce Strong
and etc.) are not allowed to be configured as password.
Password
2. Password for extension registration, Web GUI admin login, LDAP and
LDAP sync requires alphanumeric characters containing at least two
categories of the following, with a minimum length of 4 characters.
• Numeric digits
• Lowercase alphabet characters
• Uppercase alphabet characters
• Special characters
Enable Random 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 pre-
Disable Extension configured/configured on the UCM6200. The default setting is "No".
Range Note: While there are no issues with disabling extension ranges, system
administrators will need to manage extensions to avoid extension conflicts.
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.

Extension Ranges
• 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".

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 If "Enable Pick Extension" under zero config settings is selected, the
extension list defined in "Pick Extension Segment" will be sent out to the
device after receiving the device's request. This "Pick Extension Segment"
should be a subset of the "Pick Extensions" range here. This feature is for
the GXP series phones that support selecting extension to be provisioned
via phone's LCD.

• Auto Provision Extensions: 5000-6299

This sets the range for "Zero Config Extension Segment" which is the
extensions can be assigned on the UCM6200 to provision the end device.

• Conference Extensions: 6300-6399

• Ring Group Extensions: 6400-6499
• Queue Extensions: 6500-6599
• Voicemail Group Extensions: 6600-6699
• IVR Extensions: 7000-7100
• Dial By Name Extensions: 7101-7199
• Fax Extensions: 7200-8200

PBX Settings/RTP Settings

RTP Settings

Table 100: 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".

Configure to enable or disable RTP Checksums on RTP traffic. The default

RTP Checksums
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 is used to check the connectivity between the
two terminals, such as maintaining a NAT binding entries keep-alive agreement.
The default STUN Server is stun.ipvideotalk.com.
STUN Server

Valid format:
[(hostname | IP-address) [':' port]
The default port number is 3478 if not specified.
BFCP UDP Start Configure BFCP UDP port starting number. The default setting is 50000.
BFCP UDP End Configure BFCP UDP port ending number. The default setting is 52999.
BFCP TCP Start Configure BFCP TCP port starting number. The default setting is 53000.
BFCP TCP End Configure BFCP TCP port ending number. The default setting is 55999.
Configure TURN server address. TURN server is an enhanced version of the
TURN Server
STUN protocol and is dedicated to the processing of symmetric NAT pronblems.
TURN Server Name Configure TURN server account name.
TURN Server
Configure TURN server account password.
Passsword

Payload

The UCM6200 payload type for audio codecs and video codes can be configured here.

Table 101: 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.
Configure the payload type for G.726 if "G.721 Compatible" is disabled. The
G.726
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.265 Configure the payload type for H.264. The default setting is 114.
H.263P Configure the payload type for H.263+. The default setting is 100 103.
VP8 Configure the payload type for VP8. The default setting is 108.
Main Video FEC Configure the payload type for Main Video FEC, The default is 120.
RTP FECC Configure the payload type for RTP FECC, The default is 125

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 Configure the payload type for RTX, The default is 124.
RTX
Note: UCM supports only video RTP retransmission.

PBX Settings/Voice Prompt Customization

Record New Custom Prompt

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 243: 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.

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.

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• 8000Hz mono.
• In .mp3 or .wav format; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix.
• File size under 5M.

Figure 244: 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 245: Download All Custom Prompt

Note: The downloaded file will have a .tar extension.

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Call Failure Tone Settings

SIP Trunk Prompt Tone

Prompt Tone Settings tab has been added to the UCM to help users choose which prompt will be played by
the UCM during call failure, the following voice message responses have been added and can be set to be
played for 4XX, 5XX, and 6XX call failures:

• 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.

Figure 246: SIP Trunk Prompt Tone

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General Call Failure Tone

Moreover, users also have the possibility to customize the prompt for typical call failure reasons like (no
permission to allow outbound calls, busy lines, incorrect number dialed …Etc.).

To customize these prompts user could record and upload their own files under “’PBX Settings → Voice Prompt
→ Custom Prompts” then select each one for specific call failure case under “PBX Settings -> Prompt Tone
Settings → General Call Prompt Tone” page as shown on the following figure:

Figure 247: General call Failure Prompts

PBX Settings/Jitter Buffer

Table 102: Internal Options/Jitter Buffer

SIP Jitter Buffer

Select to enable jitter buffer on the sending side of the SIP channel. The default
Enable Jitter Buffer
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

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 The size is adjusted automatically and the maximum value equals to the
value of "Max Jitter Buffer".

PBX Settings/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.

Figure 248: 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.

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 Figure 249: Recordings Storage Prompt Information

Click on “OK” to continue. The users will be prompted a new dialog to select the categories for the files to be
copied over.

Figure 250: Recording Storage Category

On the UCM6200, recording files are generated and exist in 3 categories: normal call recording files, conference
recording files, and call queue recording files. Therefore, users have the following options when select the
categories to copy the files to the external device:

- Recording Files: Copy the normal recording files to the external device.
- Conference: Copy the conference recording files to the external device.
- Queue: Copy the call queue recording files to the external device.
- All: Copy all recording files to the external device.

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PBX Settings/NAS

UCM supports saving call recordings and backing up/restoring system backups to a NAS server. The following
options are available:

Table 103: NAS Settings

Enable Toggles the NAS recording functionality.

Configure the Domain or IP address of the NAS server.

Host
Note: Currently, only IP addresses are supported in the Host/IP field.

Share Name Specify the name of the shared folder.

Username Specify the account username to access the NAS server.

Password Configure the account password to access the NAS server.

If configured correctly, the Status field will show “Mounted”, and the newly
added NAS server will be shown on the Mounted Netdisk List. Additionally, the
Status
NAS will appear as a selectable storage option in the PBX Settings→Recording
Storage page and CDR→Recording Files page.

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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 104: SIP Settings/General

Configure the host name or domain name for the UCM6510. Realms MUST be
Realm For Digest
globally unique according to RFC3261. The default setting is grandstream.
Authentication
Note: Both FQDN and IP address format are supported in this field.
Bind UDP Port Configure the UDP port used for SIP. The default setting is 5060.
Configure the IPv4 address to bind to. The default setting is 0.0.0.0, which means
Bind IPv4 Address
binding to all addresses.
Configure the IPv6 address to bind to. The default is : "[::]" and it means to bind
Bind IPv6 Address
to all IP addresses.
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.
If disabled, the UCM will not forward the diversion header.
Enable Diversion
Header Note: Diversion header will be included for Forward and transfer when the option
is enabled.
If enabled, collect calls will be blocked.

Block Collect Calls Note: Collect calls are indicated by the header "P-Asserted-Service-Info: service-
code=Backward Collect Call, P-Asserted-Service-Info: service-code=Collect
Call".

SIP Settings/MISC

Table 105: 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.
Register Attempts The default setting is 0, which means the UCM6200 will keep trying until the
server side accepts the registration request.

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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".
If enabled, when rejecting an incoming INVITE or REGISTER request, the
UCM6200 will always reject with "401 Unauthorized" instead of notifying the
Reject Non-Matching
requester whether there is a matching user or peer for the request. This reduces
INVITE
the ability of an attacker to scan for valid SIP usernames. The default setting is
"No".
SDP Attribute Passthrough
Enable Attribute If enabled, and UCM receives a call that contains unknown FEC/FECC/FBCP
Passthrough attributes, they will be passed through the UCM unmodified.
Early Media
Enable Use Final SDP If enabled, call negotiation will use final response SDP.
Blind Transfer
If enabled, the UCM will call back to the transferrer when blind transfer fails (due
Allow callback when
to the destination being busy or not answering).
blind transfer fails
Note: This feature applies only to internal calls.
Configure the amount of time in seconds that the transferred party will wait for
Blind transfer timeout the destination to answer before being redirected back to the transferrer. Default
is 60 seconds.
DNS
This option affects the DNS query only during Calls. When you choose A&AAAA,
UCM will do both A and AAAA type DNS query; when you chose A, UCM will only
DNS mode
do A type DNS query; and when you chose AAAA, UCM will only do AAAA type
DNS query.
Hold
Configure the UCM to forward HOLD requests instead of processing holds
internally. This serves to meet the standards set by some providers that require
Forward HOLD HOLD requests to be passed along from endpoint to endpoint. This option is
Requests disabled by default.
Note: Enabling this option may cause hold retrieval issues and MOH to not be
heard.

SIP Settings/Session Timer

Table 106: SIP Settings/Session Timer

Force Timer If checked, always request and run session timer.

Timer If checked, run session timer only when requested by other UA.
Session Expire Configure the maximum session refresh interval (in seconds).

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 The default setting is 1800.
Configure the minimum session refresh interval (in seconds).
Min SE
The default setting is 90.

SIP Settings/TCP and TLS

Table 107: SIP Settings/TCP and TLS

Configure to allow incoming TCP connections with the UCM6200. The default
TCP Enable
setting is "No".
Configure the IP address for TCP server to bind to. 0.0.0.0 means binding to all
TCP Bind Address
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

If enabled, the TLS server's certificate will not be verified when acting as a client.
TLS Do Not Verify
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.
TLS Cert This file will be renamed as "TLS.pem" automatically.

Note:
The size of the uploaded certificate file must be under 2MB.
This file must be named with the CA subject name hash value. It contains CA's
(Certificate Authority) public key, which is used to verify the accessed servers.
TLS CA Cert
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 108: 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 a hostname is used, it will be looked up only
once.

If enabled, the SDP connection will use the IP address resolved from the external
Use IP address in SDP
host.

Configure the externally mapped TCP port when the UCM6200 is behind a static
External TCP Port
NAT or PAT.

Configures the externally mapped TLS port when UCM6200 is behind a static
External TLS Port
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 109: 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
Configure the default duration (in seconds) of incoming/outgoing registration.
Incoming/Outgoing
The default setting is 120.
Registration Time
Max
Configure the maximum allowed duration (in seconds) of incoming registration
Registration/Subscrip
and subscription requests. Default value is 3600.
tion Time
Min
Configure the minimum allowed duration (in seconds) of incoming registration
Registration/Subscrip
and subscription requests. Default value is 60.
tion Time
Select to enable relaxed DTMF handling, This may cause audio to be mistaken
Enable Relaxed DTMF
for DTMF. The default setting is "No".
Select DTMF mode to send DTMF. The default setting is RFC4733. If "Info" is
DTMF Mode selected, SIP INFO message will be used. If "Inband" is selected, 64-kbit codec
PCMU and PCMA are required. When "Auto" is selected, "RFC4733" will be used

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 if offered. Otherwise "Inband" will be used. The default setting is "RFC4733".

During an active call, if there is no RTP activity before the configured timeout,
the call will be terminated. The default setting is no timeout.
RTP Timeout
Note: This setting does not apply to calls on hold.

When the call is on hold, if there is no RTP activity before the configured
RTP Hold Timeout 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”.
Configure whether the Remote-Party-ID should be trusted. Default setting is
Trust Remote Party ID
"No".
Configure whether the Remote-Party-ID should be sent or not. The default
Send Remote Party ID
setting is "No".
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 • No: The UCM6200 will send 180 Ringing if 183 Session Progress has not
Ringing been sent yet. If audio path is established already with 183 then send in-
band ringing.
• Never: Whenever ringing occurs, the UCM6200 will send 180 Ringing as
long as 200OK has not been set yet. Inband ringing will not be generated
even the end point device is not working properly.
Configure the user agent string for the UCM6200.
Server User Agent Note: The value configured in the Server User Agent Value field will replace the
whole User-Agent header value instead of just the “UCM6xxx” part.
Send Compact SIP
If enabled, compact SIP headers will be sent. The default setting is "No".
Headers

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

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and give the option to open the door via softkey.

Figure 251: Transparent Call-Info

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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 110: 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 111: 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 112: IAX Settings/Static Defense

Enter a single IP address (e.g., 11.11.11.11) or a range of IP addresses

Call Token Optional
(11.11.11.11/22.22.22.22) for which call token validation is not required.

Max Call Numbers Configure the maximum number of calls allowed for a single IP address.

Max Unvalidated Call

Configure the maximum number of Unvalidated calls for all IP addresses.
Numbers

Call Number Limits 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

IP or IP Range
(11.11.11.11/22.22.22.22) to be considered for call number limits.

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 INTERFACE SETTINGS

Analog Hardware

The analog hardware (FXS port and FXO port) on the UCM6200 will be listed in this page. Click on to edit
signaling preference for FXS port or configure ACIM settings for FXO port.
Select "Loop Start" or "Kewl Start" for each FXS port. And then click on "Update" to save the change.

Figure 252: FXS Ports Signaling Preference

For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for each
port. Or users could click on "Detect" and choose the detection algorithm, two algorithms exist (ERL, Pr) for the
UCM6200 to automatically detect the ACIM value. The detecting value will be automatically filled into the settings.

Figure 253: FXO Ports ACIM Settings

Table 113: 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".

Configure to increase the ringing speed to 25HZ. This option can be used with
Fast Ringer
"Low Power" option. The default setting is "Normal".

Configure the peak voltage up to 50V during "Fast Ringer" operation. This
Low Power
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
setting is "FSK".
FXS MWI Mode
• FSK: Frequency Shift Key Indicator
• NEON: Light Neon Bulb Indicator.
FXO Frequency Allows users to adjust the tolerance of the FXO ringing frequency. 63Hz is
Tolerance considered the standard value and is selected by default.

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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 254: 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.
DTMF Start Threshold: Indicates the minimum number of times a single DTMF must be detected in a signal
before it is considered valid.
DTMF End Threshold: Indicates the minimum number of times an end signal for a single DTMF number must
be detected before it is considered valid.

Normally, DAHDI settings should be kept default and should be adjusted only when users encounter analog
trunk/Fax-related issues.

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 GDMS SETTINGS

UCM can synchronize its SIP accounts to GDMS cloud management system.
http://www.grandstream.com/products/device-management/gdms

To get started, log into your GDMS account and navigate to the System→API Developer page. Click on the
Enable API Developer Mode button if it has not been enabled yet.

Figure 255: GDMS Developer Mode Button

After clicking on it, the API ID and Secret Key will be displayed. Note down these credentials.
On the UCM, navigate to System Settings→GDMS Settings. Check the Enable option if it has not been toggled
on. Enter your GDMS account credentials and the API developer credentials.
For the Account field, enter either your account username or email address. Once all the information has been
entered, click on the Authenticate button to connect to GDMS.

Figure 256: GDMS API Credentials

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 Figure 257: GDMS Settings

Toggles GDMS support. If enabled, all SIP extensions on the UCM will be
Enable
automatically synced to GDMS.

Select your GDMS server region:

Server Region • US region
• EU region

Account Email/Username The account email/username for GDMS authentication.

Password The password for GDMS authentication.

API ID from GDMS account.

API ID
Refer to [Figure 256: GDMS API Credentials].

API secret key from GDMS account.

Secret Key
Refer to [Figure 256: GDMS API Credentials].

Connection Status Provides the connection status once authenticated.

Name of the organization in GDMS that the UCM SIP server and its
Organization
extensions will be under.

Authenticate Sends the authentication request to GDMS with the entered credentials.

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If the authentication is successful, you will now be able to select an organization on GDMS to synchronize all
UCM SIP accounts to. After selecting, saving and applying changes, the UCM will then start the syncing process.
Once the initial sync is complete, you will now be able see the UCM and its extensions on GDMS, which will
have an orange label with the words “UCM”. Any future creation, deletion, or modification of SIP accounts will
automatically be synchronized to GDMS.

Figure 258: UCM on GDMS

Figure 259: UCM SIP Extensions on GDMS

Any future creation, modification, and deletion of the UCM’s SIP extensions will automatically be synchronized
to GDMS.
Note: UCM extensions’ names currently cannot be synchronized.

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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. More methods are also supported to provide better
integration with 3rd party systems.

Before accessing the API, the administrators need to enable the HTTPS API and configure the
access/authentication information on the UCM6200 first under Value-added Features→API Configuration.
The API configuration parameters are available for HTTPS API Settings (New), HTTPS API Settings (Old), CDR
Real-time Output Settings & “Upload Prompts User Configuration”.

HTTPS API (New)

Starting from firmware 1.0.20.17, UCM6200 supports new HTTPS API interface to query, edit PBX settings and
implement multiple call functions on another server connected to it via API. PBX will actively send system reports
and call reports to this other server. Additionally, legacy CDR API, REC API and PMS API are supported.

The table below lists configuration parameters for HTTPS API.

Table 114: API Configuration Parameters

HTTPS API Settings (New)

Enable Enable/Disable API. The default setting is disabled.

Username Configure the username for API Authentication.

Password Configure the password for API Authentication.

If enabled, 3rd party applications will be able to manage inbound calls via API
Call Control actions. acceptCall will accept incoming calls while refuseCall will reject them. If
no actions are done within 10 seconds, calls will automatically be accepted.
Note: HTTPS API uses web interface port (default is 8089).

The table below lists new HTTPS API supported methods.

Table 115: New API Supported Queries

getSystemStatus addInboundRoute listPaginggroup

getSystemGeneralStatus getInboundRoute addPaginggroup

listAccount updateInboundRoute getPaginggroup

getSIPAccount deleteInboundRoute updatePaginggroup

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 updateSIPAccount playPromptByOrg deletePaginggroup

listVoIPTrunk listBridgedChannels MulticastPaging

addSIPTrunk listUnBridgedChannels MulticastPagingHangup

getSIPTrunk Hangup listIVR

updateSIPTrunk Callbarge addIVR

deleteSIPTrunk listQueue getIVR

listOutboundRoute getQueue updateIVR

addOutboundRoute updateQueue deleteIVR

getOutboundRoute addQueue cdrapi

updateOutboundRoute deleteQueue recapi

deleteOutboundRoute loginLogoffQueueAgent pmsapi

listInboundRoute pauseUnpauseQueueAgent queueapi

listDigitalTrunk listAnalogTrunk getAnalogTrunk

deleteAnalogTrunk updateAnalogTrunk addAnalogTrunk

addSLATrunk updateSLATrunk addDigitalTrunk

updateDigitalTrunk deletedigitalTrunk getDigitalTrunk

For more details, please refer to online how-to guide available in our website.

HTTPS API (Old)

Table 116: API Configuration Parameters (Old)

HTTPS API Settings (Old)

Basic Settings
Enable Enable/Disable API. The default setting is disabled.

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
TLS Bind Address address must match the common name (host name) in the certificate so that the
TLS socket will not bind to multiple IP addresses.
The default setting is 0.0.0.0:8443.

Username Configure the username for TLS authentication.

Password Configure the password for TLS authentication.

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 Specify a list of IP addresses permitted to use the API. This creates an API-specific
access control list. Multiple entries are allowed.
For example, "192.168.40.3/255.255.255.255" denies access from all IP addresses
Permitted IP(s)
except 192.168.40.3.
By default, this is blank, which indicates that no IP addresses are allowed to use
this API.
Other Settings

Upload TLS private key. The size of the key file must be under 2MB. This file will be
TLS Private Key
renamed as 'private.pem' automatically.

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
TLS Cert
"certificate.pem" automatically. It contains private key for the client and signed
certificate for the server.

API Module

CDR API Enable/disable CDR API module.

REC API Enable/disable REC API module.

PMS API Enable/disable PMS API module.

For more details on CDR API (Access to Call Detail Records), REC API (Access to Call Recording Files) and
PMS API, please refer the document in the link here:
• CDR API
• REC API
• PMS API

CDR Real-time Output Settings

CDR Real-time output feature allows to automatically send CDR records once available (after the call is
terminated) to a specified server address.

Table 117: CDR Real-time Output Settings

CDR Real-time Output Settings

Enables real-time CDR output module. This module connects to selected IP

Enable
addresses and ports and posts CDR strings as soon as it is available.

Server Address CDR server IP address

Port CDR server IP port

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 Delivery Method Choose either TCP protocol or HTTP/HTTPS protocol

Format Choose either XML or JSON for CDR format

Username Enter the Username for authentication.

Password Enter the Password for authentication.

For more details, refer to online how-to guide available at:

http://www.grandstream.com/sites/default/files/Resources/CDR_Real-time_Output_Feature_Guide.pdf

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 CTI SERVER

UCM supports CTI server capabilities which are designed to be a part of the CTI solution suite provided by
Grandstream, which involves the GXP17XX series, GXP21XX series and the GS Affinity application.

By default, the UCM listens to port 8888 for connection requests from the GS Affinity application. This will allow
the UCM to interact with, modify, and execute requests by the application such as setting call forwarding 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 260: 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 is 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 support the following CRM API: SugarCRM, vTigerCRM, ZohoCRM and Salesforce CRM.
This support allows users to quickly create and modify contacts, leads, and accounts in the CRM from calls
made through the UCM.

Note: Starting firmware 1.0.17.16, emergency calls will not be logged into CRM servers.

SugarCRM

Configuration page of the SugarCRM can be accessed via admin login, on the UCM webGUI→Value-added
Features→CRM.

Figure 261: SugarCRM Basic Settings

1. Select “SugarCRM” from the CRM System Dropdown in order to use SugarCRM.

Table 118: SugarCRM Settings

CRM System Select a CRM system from the dropdown menu, four CRM systems
are available: SugarCRM, vTigerCRM, ZohoCRM (legacy v1 API),
ZohoCRM (v2 API), Salesforce or ACT! CRM.
CRM Server Address Enter the IP address of the CRM server.
Add Unknown Number Add the new number to this module if it cannot be found in the

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 selected module.
Contact Lookups
Select from the “Available” list of lookups and press to
select where the UCM can perform the lookups on the CRM tables,
Leads, Accounts, and Contacts.

Once settings on admin access are configured:

2. Click on and .
3. Logout from admin access.
4. Login to the UCM as user and navigate under “User 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 262: CRM User Settings

vTigerCRM

Configuration page of the vTigerCRM can be accessed via admin login, on the UCM webGUI→Value-added
Features→CRM.

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 Figure 263: vTigerCRM Basic Settings

1. Select “vTigerCRM” from the CRM System Dropdown in order to use vTigerCRM.

Table 119: vTigerCRM Settings

CRM System Select a CRM system from the dropdown menu, four CRM systems
are available: SugarCRM, vTigerCRM, ZohoCRM (legacy v1 API),
ZohoCRM (v2 API), Salesforce or ACT! CRM.
CRM Server Address Enter the IP address of the CRM server.
Add Unknown Number Add the new number to this module if it cannot be found in the
selected module.
Contact Lookups
Select from the “Available” list of lookups and press to
select where the UCM can perform the lookups on the CRM tables,
Leads, Organizations, and Contacts.

Once settings on admin access are configured:

2. Click on and .
3. Logout from admin access.
4. Login to the UCM as user and navigate under “User 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.

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 Figure 264: CRM User Settings

ZohoCRM

Configuration page of the ZohoCRM v1 and ZohoCRM v2 can be both accessed via admin login, on the UCM
Web GUI→Value-added Features→CRM.

Figure 265: ZohoCRM Basic Settings

1. Select “ZohoCRM (v2 API)” from the CRM System Dropdown in order to use ZohoCRM.

Note: Zoho CRM (legacy v1 API) will no longer be supported after 2019. Please use Zoho CRM (v2
API).

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 Table 120: ZohoCRM Settings

CRM System Select CRM system from the dropdown menu, four CRM systems
are available: SugarCRM, vTigerCRM, ZohoCRM (legacy v1 API),
ZohoCRM (v2 API), Salesforce or ACT! CRM.
CRM Server Address Select Zoho CRM URL from the list. Available options are:
• https://www.zohoapis.com
• https://www.zohoapis.com.cn
• https://www.zohoapis.eu
Add Unknown Number Add the new number to this module if it cannot be found in the
selected module.
Contact Lookups
Select from the “Available” list of lookups and press to
select where the UCM can perform the lookups on the CRM tables,
Leads, Accounts, and Contacts.

Once settings on admin access are configured:

2. Click on and .
3. Logout from admin access.
4. Login to the UCM as user and navigate under “User 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

ZohoCRM features.

Figure 266: CRM User Settings

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Salesforce CRM

Configuration page of the Salesforce CRM can be accessed via admin login, on the UCM Web GUI→Value-
added Features→CRM”.

Figure 267: Salesforce Basic Settings

1. Select “Salesforce” from the CRM System Dropdown in order to use Salesforce CRM.

Table 121: Salesforce Settings

CRM System Select a CRM system from the dropdown menu, four CRM systems
are available: SugarCRM, vTigerCRM, ZohoCRM (legacy v1 API),
ZohoCRM (v2 API), Salesforce or ACT! CRM.
Add Unknown Number Add the new number to this module if it cannot be found in the
selected module.
Contact Lookups
Select from the “Available” list of lookups and press to
select where the UCM can perform the lookups on the CRM tables,
Leads, Accounts, and Contacts.

Once settings on admin access are configured:

2. Click on and .
3. Logout from admin access.
4. Login to the UCM as user and navigate under “User 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 268: Salesforce User Settings

ACT! CRM

Configuration page of the ACT! CRM can be accessed via admin login, on the UCM Web GUI→Value-added
Features→CRM”.

The configuration steps of the ACT! CRM are as follows:

1. Navigate to Value-Added Features->CRM and select the “ACT! CRM” option.

Figure 269: Enabling ACT! CRM

2. Log into the UCM as a regular user and navigate to Value-Added Features → CRM User Settings and
check “Enable CRM” option and enter the username and password, which will be the ACT! CRM
account’s API Key and Developer Key, respectively. To obtain these, please refer to the ACT! CRM API
developer’s guide here: https://mycloud.act.com/act/Help

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 Figure 270: Enabling CRM on the User Portal

Note: For more information on the ACT! CRM integration, please refer to the ACT! CRM documentation on our
website.

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 PMS INTEGRATION

UCM6200 supports Hotel Property Management System PMS, including check-in/check-out services, wakeup
calls, room status, Do Not Disturb which provide an ease of management for hotel applications. This feature can
be found on Web GUI→Value-added Features→PMS.
Note: The PMS integration on UCM is currently supported only with one of the three following solutions.

The PMS module built-in the UCM supports the following features based on each solution:

Table 122: PMS Supported Features

Feature Mitel HMobile HSC

Check-In ✓ ✓ ✗

Check-out ✓ ✓ ✗

Wake-up Call ✓ ✓ ✗

Name Change ✓ ✗ ✓

Update ✗ ✓ ✗

Set Credit ✓ ✗ ✗

Set Station Restriction ✓ ✗ ✓

Room Status ✗ ✓ ✗

Room Move ✗ ✓ ✗

Do Not Disturb ✗ ✓ ✓

Mini Bar ✗ ✓ ✗

MSG ✗ ✓ ✗

MWI ✗ ✗ ✓

Unconditional Call Forward ✗ ✗ ✓

HMobile PMS Connector

In this mode, the system can be divided into three parts:

• PMS (Property Management System)

• PMSI (Property Management System Interface)

• PBX

Grandstream UCM6XXX series have integrated HMobile Connect PMSI which supports a large variety of PMS
software providing following hospitality features: Check-in, Check-out, set Room Status, Wake-up call and more.

The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and
PMS software, which is done through a middleware system (HMobile Connect) acting as interface between both
parties.

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 PMS Software

PMSI Middleware
(HMobile Connect)

Grandstream UCM

Figure 271: UCM & PMS interaction

HSC PMS

In this mode, the system can be divided into two parts:

• PMS (Property Management System)

• PBX

Grandstream UCM6XXX series have integrated HSC PMS providing following features:

• Changing Display Name

• Set Station Restriction
• Call forwarding
• DND
• Name Change
• MWI

The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and
PMS software (HSC). The communication between both parties is direct with no middleware.

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 HSC PMS Software

Grandstream UCM

Figure 272: UCM & HSC PMS interaction

Mitel PMS

In this mode, the system can be divided into two parts:

• PMS (Property Management System)

• PBX

Grandstream UCM6XXX series have integrated Mitel PMS providing following hospitality features: Check-in,
Check-out, set Room Status, Wake-up call and more.
The following figure illustrates the communication flow between the PBX (Grandstream UCM6xxx Series) and
PMS software (Mitel). The communication between both parties is direct with no middleware.

Mitel PMS Software

Grandstream UCM

Figure 273: UCM & Mitel PMS interaction

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PMS API

The PMS API allows users to use their own middleware to work with PMS systems instead of currently supported
integrations.

Additionally, this API allows access to read and modify certain UCM parameters that current supported PMS
integrations cannot. To use this, users must first enable and configure the HTTPS API settings.

On the HTTPS API settings, permitted IP addresses must be configured. Otherwise, the API will be inaccessible.

On the “Old” HTTPS API settings, permitted IP addresses must be configured. Otherwise, the API will be
inaccessible. Make sure to check “Enable PMS API”.

Figure 274: Enable PMSAPI

On the “New” HTTPS API, users only need to enable the HTTPS API, which has PMS API functionality built in.

For more details, regarding Old/New HTTPS API, please refer to [API CONFIGURATION].
For more details, please refer to online PMS API Guide.

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Connecting to PMS

On the UCM WebGUI→Value-added Features→PMS→Basic Settings” set the connection information for the
PMS platform.

Table 123: PMS Basic Settings

Field Description
Users can select the desired PMS module from the drop-down list.
• Hmobile.
PMS Module • Mitel.
• HSC.
• PMS API

Prompt used when answering the wakeup calls it can be customized

Wake Up Prompt
from “PBX Settings→Voice Prompt→Custom Prompt.

PMS URL Enter the PMS system URL.

If using “Hmobile” PMS Module only.

UCM Port Enter

If the Port used by the PMS system. Default is 8081.

Username Enter the Username to connect to the PMS system.

If using “Hmobile” or “HSC” PMS Module only.

Password Enter the password to connect to the PMS system.

If using “Hmobile” or “HSC” PMS Module only.
Site Enter the site to connect to the PMS system.
If using “Hmobile” PMS Module only.

Back Up Voicemail If enabled, this option allows backing up voicemail recordings to external
Recordings storage after check-out. Voicemail can be backed up to SD card, USB
disk, NAS, or an SFTP server

Email address Configure the email address to send the backup to.

In order to use some PMS features please activate the feature code associated under “Call Features→Feature
Codes”

• Update PMS Room Status

• PMS Wake Up Service

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PMS Features

Room Status

User can create Rooms by clicking on , the following Figure will be displayed then.

Figure 275: Create New Room

Click “Save” to create the new room, the fields above can be configured from the PMS platform, once set the
following screen will be shown:

Figure 276: Room Status

User can create a batch of rooms as well by clicking on .

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The following windows will appear:

Figure 277: Add batch rooms

Wake Up Service

To create a New Wake up service, user can click on .

The following windows will appear:

Figure 278: Create New Wake Up Service

Table 124: PMS Wake up Service

Field Description
Room Number Select the room to create the wakeup service for.
Time Set the time of the wakeup call
Action Status Status of the call:
• Programmed: the call is scheduled for the time set
• Cancelled: the call is canceled
Executed: the wakeup call has been sent.
Note: Editing an already executed wakeup service will automatically change
the service’s status to “Programmed”.
Type Single: The call will be made once at the specified time.

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 • Daily: The call will be repeated every day at the specified time.

Once the call is made on the time specified, the following figure show the status of the wakeup call.

Figure 279: Wakeup Call executed

Mini Bar

In order to create a new mini bar, click on under UCM webGUI→Value-added

Features→PMS→Mini Bar.

The following windows will appear:

Figure 280: Create New Mini Bar

Table 125: 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 If enabled, please separate the goods’ codes by*.
Billing

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In order to create a new maid, click on under UCM webGUI→Value-added
Features→PMS→Mini Bar.

The following windows will appear:

Figure 281: Create New Maid

Table 126: 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.

In order to create a new consumer goods, click on under UCM webGUI→Value-

added Features→PMS→Mini Bar.

The following windows will appear:

Figure 282: Create New Consumer Goods

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 Code Enter the Goods Code.
Name Enter the Name of the Goods

The Minibar page displays as:

Figure 283: Mini Bar

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 WAKEUP SERVICE

The Wakeup service can be used to schedule a reminder/wakeup calls to extensions.

There are three ways to set up Wakeup Service:

• Using admin portal

• Using user portal

• Using feature code

Wakeup Service using Admin Login

1. Log in 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 284: Create New Wakeup Service

3. Fill out the required fields and select the members to add to the wakeup group.

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 Table 127: Wakeup Service

Enable Wakeup Service Enable Wakeup service.

Name Enter a name (up to 64 characters) to identify the wakeup service.

Prompt Select the prompt to play upon answering the wakeup call.

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 to send the wakeup call.

Members Select the extensions to send the wakeup call to.

4. Click and to apply the changes.

A wakeup service entry is created. The UCM will send a wakeup call to every extension in the member list at
the scheduled date and time.

Note: the wakeup service has the following limitation on how many members can be added depending on UCM
model.

Table 128: Max Wakeup Members

UCM Model Max members in a Wakeup Service

UCM6202 50

UCM6204 50

UCM6208 100

UCM6510 100

Wakeup Service from User Portal

1. Log into the UCM user portal.

2. Wakeup service can be found under “Value-added Features→Wakeup Service”, click on

to create a new wakeup service.

3. Configure the Name, Prompt, Date and Time for the wakeup call.

4. Click and to apply the changes.

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Wakeup Service using Feature Code

• Log into the UCM admin portal.

• Enable “Wakeup Service” from the WebGUI under the Call Features->Feature Codes->Feature Codes
page.

Figure 285: Wakeup Service Feature Code

• Click and to apply the changes.

• Dial the Wakeup Service feature code (*36 by default) to access to the UCM wakeup service.
• Users have the choice to set a wakeup service for tomorrow or specify another day when adding a
wakeup service.

• A wakeup service entry is created. The UCM will send a wakeup call at the scheduled date and time.

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 ANNOUNCEMENTS CENTER

Starting from firmware 1.0.2.7, UCM supports Announcement Center functionality. By dialing a code
associated with a custom prompt along with a configured Announcement Group number, users can quickly
send audio prompts to specified groups of extensions.

Figure 286: Announcements Center

Announcements Center Settings

Table 129: Announcements Center Settings

Name Configure a name for the Announcements.

Enter a code number to associate the audio prompt with. This code will be
used in combination with the Group Number. Example: If the code is 55, and
the group number is 666, dialing 66555 will send the prompt associated with
Code
code 55 to all the members of Group 666.
Note: The combined number must not conflict with any existing extension on
the UCM.
Configures the custom prompt to play to announcement group members.
Custom Prompt Prompts can be directly uploaded from this page. Note: Prompt filenames
cannot exceed 100 characters.
Configure the ring timeout for the group members. The default value is 30
Ring Timeout
seconds.

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Group Settings

Table 130: Group Settings

Configure a name for the announcement group.

Name
Note: Name cannot exceed 64 characters
Configure the group number. Please see recommendation for Code Number for
Number
reference.

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. Configure a name and number for the group.

3. Create a group number which is used with code to send voice message.

4. Select the extensions that will be ion the group.

Figure 287: Announcements Center Group Configuration

Here, Group Test's number is 666 and has extensions 1000, 1001, and 1002 as members of the group.

1. Click to create a new Announcement Center.

2. Configure the Announcement Center name and code to associate with the custom prompt.

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3. Select the audio prompt that will be associated with the configured code. Users can directly upload audio
prompts to use from this page.

Figure 288: Announcements Center Code Configuration

Sending 55 (code) + 666 (group number) will send an announcement with the associated audio prompt to all
the members of Group Test.

Figure 289: Announcements Center Example

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 QUEUEMETRICS INTEGRATION

The UCM now supports QueueMetrics, highly scalable monitoring and reporting suite that addresses the needs
of thousands of call centers worldwide and offers a broad range of benefits and services.

UCM currently supports the following features with QueueMetrics: Agent login, Agent Logoff, Realtime
Monitoring-Pausing, Realtime Monitoring-Barging, Realtime Monitoring-Transferring, Realtime Monitoring-End
calls, Generating Performance Report-Quick and Agent today.

API Configuration Parameters

Configuration page of the QueueMetrics can be accessed via admin login, on the UCM Web GUI→Value-added
Features→QueueMetrics.

To Integrate QueueMetrics with the UCM, please follow the steps below:

1. Enable the option QueueMetrics Integration.

2. Enter the QueueMetrics URL provided.

3. Set the Username and the account code to the account name and password provided from QueueMetrics.

4. Click on Save and Apply Changes.

Figure 290: QueueMetrics configuration

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 Table 131: QueueMetrics Configuration Parameters

QueueMetrics

Enable QueueMetrics
Enable/Disable QueueMetrics Integration.
Integration

QueueMetrics URL Configure the URL for QueueMetrics Integration.

Username Configure the username for QueueMetrics authentication.

Account Code Configure the password for QueueMetrics authentication.

Partition Configure Data storage partition identifier.

For more details, please refer to online guide:

http://www.grandstream.com/sites/default/files/Resources/QueueMetrics_integration.pdf

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 STATUS AND REPORTING

PBX Status

The UCM Dashboard page under System Status->Dashboard provides an real-time overview of the system's
trunks, extensions, queues, conference rooms, interface status, digital channels, and parking lots.

Figure 291: Status→PBX Status

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Trunks

Users can see the status of all configured trunks.

Figure 292: Trunk Status

Table 132: 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:
• Analog
Type
• 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 293: Extension Status

Table 133: Extension Status

Display extension number (including feature code). The color indicator has the
following definitions.
• Green: Free
Status
• Blue: Ringing
• Yellow: In Use
• Grey: Unavailable
Presence Status Display the presence status of the extension.
Extension Display the extension number.
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.
SIP Account email notification has not been sent.
Email
SIP Account email notification has been sent.

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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 294: UCM6204 Interfaces Status

Table 134: 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 135: 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 136: 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.

Speed Network Port Speed

Duplex Mode Shows the Duplex Mode (Full/Half Duplex)

Space Usage

Users could access the storage usage information from Web GUI→System Status→Dashboard→Space
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 295: 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.

Figure 296: 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 specified
email addresses.

Alert Events List

The system alert events list can be found under Web GUI→Maintenance→System Events. The following
events are currently supported on the UCM. Alerts for these events will be generated if abnormalities occur.

Table 137: Alert Events

Action index Alert Events

1 Disk Usage

2 Modify Super Admin Password

3 Memory Usage

4 System Reboot

5 System Update

6 System Crash

7 Register SIP failed

8 Register SIP trunk failed

9 Restore Config

10 User login success

11 User login failed

12 SIP Internal Call Failure

13 SIP Outgoing Call through Trunk Failure

14 Fail2ban Blocking

15 SIP Lost Registration

16 SIP Peer Trunk Status

17 User Login Banned

18 External Disk Usage

19 HA failure warning

20 Emergency Calls

21 The CDR database is corrupted

22 NAS

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 23 Data Sync Backup

24 External Disk Status

Click on to configure the parameters for each event. See examples below.
1. Disk Usage

Figure 297: System Events→Alert Events Lists: Disk Usage

• 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 298: 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.

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3. Memory Usage

Figure 299: 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.

4. System Crash

Figure 300: 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.

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 Figure 301: System Events→Alert Log

User could also filter alert logs by selecting a certain event category, type of alert log, and/or specifying a certain

time period. The matching results will be displayed after clicking on . Alert logs are classified into two

types by the system:

1. Generate Alert: Generated when alert events happen, for example, alert logs for disk usage exceeding the
alert threshold.

2. Restore to Normal: Generated when alert events being cleared, for example, logs for disk usage dropping
back below the alert threshold.

User could filter out alert logs of “Generate Alert” or “Restore to Normal” by specifying the type according to
need. The following figure shows an example of filtering out alert logs of type of “Restore to Normal”.

Figure 302: Filter for Alert Log

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Alert Contact

This feature allows the administrator to be notified when one of the Alert events mentioned above happens.
Users could add administrator's Email address under Web GUI→Maintenance→System Events→Alert
Contact to send the alert notification to an email (Up to 10 Email addresses can be added) or also specify an
HTTP server where to send this alert.

Table 138: Alert Contact

Super Admin Configure the email addresses to send alert notifications to.
Email Up to 10 email addresses can be added.
Admin Email Configure the email addresses to send alert notifications to.
Up to 10 email addresses can be added.
Email Template Please refer to section Email Templates

Protocol Protocol used to communicate with the server. HTTP or HTTPS.

Default one is HTTP.
HTTP Server The IP address or FQDN of the HTTP/HTTPS server.
HTTP Server Port HTTP/HTTPS port
Warning Template Customize the template used for system warnings.
By default:
{"action":"${ACTION}","mac":"${MAC}","content":"${WARNING_MSG}"}
Notification Customize the notification template to receive relevant alert information.
Template By default:
{"action":"${ACTION}","cpu":"${CPU_USED}","memery":"${MEM_USED}","disk":"${
DISK_USED}","external_disk":"${EXTERNAL_DISK_USED}"}
Note: The notification message with “action:0” will be sent periodically if Notification
Interval is set.
Notification Modifies the frequency at which notifications are sent in seconds.
Interval No notifications will be sent if the value is "0". Default value: 20
Template ${MAC} : MAC Address
Variables ${WARNING_MSG} : Warning message
${TIME} : Current System Time
${CPU_USED} : CPU Usage
${MEM_USED} : Memory Usage
${ACTION} : Message Type. Refer to [Table 137: Alert Events]
${DISK_USED} : Disk Usage
${EXTERNAL_DISK_USED} : Disk Usage

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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.

Important Note: Starting from fw 1.0.15.16, the UCM has CDR separation and it will display only CDR for the
current month. To view data of the previous months, users will need to apply a filter with a specified date range.

Figure 303: CDR Filter

Table 139: CDR Filter Criteria

Call Type
• Inbound calls: Inbound calls are calls originating 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 and not sent over a trunk.

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 External calls: External calls are calls sent from one trunk to another trunk and 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
Filter out calls originating from the selected trunks.
Name
Destination Trunk
Filter out calls that were sent to the selected trunks.
Name
Action Type Filter calls based on 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
• Emergency Call
• Emergency Notify
• SCA
Extension Group Filters calls made to the selected Extension Group.
Account Code Filter calls with the specified Account Code. If Outbound Routes->PIN Groups-
>Record in CDR is enabled, Account Code information will be displayed in the CDR
if applicable.
Start Time Specify the earliest date and time of CDR to filter.
End Time Specify the latest date and time of CDR to filter.

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Caller Number Enter the caller number of the CDR to filter. Patterns can be used to filter calls made
from groups of similar numbers.
Caller Name Enter the caller name of the CDR to filter.
Callee Number Enter the callee number of the CDR to filter.

The call report will display as the following figure shows.

Figure 304: 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:

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 Grandstream/Test

• Status
Answered, Busy, No answer or Failed.
Users could perform the following operations on the call report.

• Sort by “Start Time”

Click on the header of the column to sort the report by "Start Time". Clicking on "Start Time" again will
reverse the order.

• Download Search Results

Click on “Download Search Result(s)” to export the filtered CDR to a .CSV file.

• Download All Records

Click on “Download All Records” to export all the records to a .CSV file.

• Delete All

Click on button to remove all the call report information.

• Delete Search Result

On the bottom of the page, click on button to remove the filtered CDR

entries.
Note: When deleting CDR, a prompt will now appear asking whether to delete all recording files or not.

• Play/Download/Delete Recording File (per entry)

If the entry has audio recording file for the call, the three icons on the rightest column will be activated for
users to select. In the following picture, the second entry has audio recording file for the call.

Click on to play the recording file.

Click on to download the recording file in .wav format.

Click on to delete the recording file (the call record entry will not be deleted).

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 Figure 305: Call Report Entry with Audio Recording File

• Automatic Download CDR Records

Users can configure the UCM to periodically send the CDR to a configured email address. Click on
“Automatic Download” and configure the parameters in the dialog below.

Figure 306: Automatic Download Settings

Table 140: Automatic Download Settings

Automatic
Enable the Automatic Download feature.
Download
Delete Sent
Whether to delete the sent records or not.
Records
Automatic
Configure the Automatic Download Period
Download Period
Email Email address blacklist/whitelist for non-local contacts. Separate multiple addresses
with semicolon (;) (i.e."xxx;yyy").

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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. Users can click on the [+] icon to expand call details and view various legs of the call.

Figure 307: CDR Report

Figure 308: Detailed CDR Information

Downloaded CDR File

The downloaded CDR (.csv file) has different format from the Web GUI CDR. Here are some descriptions.

• Caller number, Callee number

"Caller number": the caller ID.

"Callee number": the callee ID.

If the "Source Channel" contains "DAHDI", this means the call is from FXO/PSTN line.

Figure 309: 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.

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• Source Channel, Dest Channel
Sample 1:

Figure 310: 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 311: Downloaded CDR File Sample - Source Channel and Dest Channel 2

"SIP" means it is a SIP call. There are three possible formats:

(a) PJSIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string
and can be ignored.
(c) PJSIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out through
the trunk.
(c) PJSIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this trunk.
The last XXXXX is a random string and can be ignored.
There are some other 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

CDR Export Customization

Users can select the data they want to see in exported CDR reports by first clicking on the Filter button on the
CDR page under CDR→CDR and selecting the desired information in the Export File Data field.

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 Figure 312: CDR Export File data

Statistics

CDR Statistics is an additional feature on the UCM6200 which provides users a visual overview of calls across
a period of time. Users can filter with different criteria to generate the statistics chart.

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 Figure 313: CDR Statistics

Table 141: CDR Statistics Filter Criteria

Trunk Type Select one of the following trunk types.

• All
• SIP Calls
• PSTN Calls
Call Type Select one or more in the following checkboxes.
• Inbound calls
• Outbound calls
• Internal calls
• External calls
• All calls
Time Range • By month (of the selected year).
• By week (of the selected year).
• By day (of the specified month for the year).
• By hour (of the specified date).
• By range. For example, 2016-01 To 2016-03.

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Recording Files

The Recording Files page lists all recordings saved via "Auto Record" or "Start/Stop Call Recording" feature
code. If external storage (e.g., NAS, SD card, USB disk, etc.) is connected, recordings will be saved there. If
no external storage is available, the recordings will be stored in UCM local storage.

Figure 314: 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 “Batch Download Recording Files” in order to download the selected recording files.

• Click on “Download All Recording Files” to download all recordings files.

• Select Either “USB Disk”, “NAS” or “Local” to show recording files stored on external or internal storage,

depending on selected storage space.

• Select whether to show call recordings, queue recordings or conference recordings.

• Click on to download the recording file in .wav format.

• Click on to delete the recording file.

• To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding column. Click
on the title again can switch the sorting mode between ascending order or descending order.

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.

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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 142: 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 will not 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 Fax Sending, Call Queue, Wakeup Service and CRM.

Users also can access their personal data files such as call recordings, fax files, and voicemail promts.

The login credentials are configured by Super Admin. The following figure shows the User Management->Edit
User page for extension 1000. Username for an extension account cannot be modified. Please note that User
Password and SIP Password are different fields and should not be mistaken for one another.

Figure 315: Edit User Information by Super Admin

The following figure is an example of logging in with an extension account.

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 Figure 316: User Portal Login

Upon successfully logging in, the following page will appear.

Note: Regular users will be logged out automatically if an admin resets their extensions.

Figure 317: User Portal Layou

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Basic Information

Under this menu, the user can configure and change his/her personal information such as first name, last
name, user password, email address, and department. Users can set extension features such as presence
status and call forwarding and view their CDR.

Personal Data

In this section, users can access and manage their personal files such as voicemail, call recordings, and fax
and set up Follow-Me.

Value-added Features

On this section, the user has access to manage and use all rich value-added features which includes:

Fax Sending

For Fax Sending configuration parameters, please refer to the [FAX Sending] section.

Call Queue

If user is a member of call queue, they can check the queue’s activity and for more information, please refer to [CALL
QUEUE] section.

Wakeup Service

This section allows users to create and enable Wakeup service and for more information, please refer to
[WAKEUP SERVICE] section.

CRM User Settings

This section allows users to enable and configure CRM connection to either SugarCRM or Salesforce.

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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 318: 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.

Figure 319: Create New User

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 Table 143: User Management→Create New User

Configure a username to identify the user which will be required in Web GUI login.
Username
Letters, digits and underscore are allowed in the username.
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 320: 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 is the only user
with “Super Admin” privilege. The Username and Privilege level cannot be changed or deleted.
- Super Admin could change its own login password on Web 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→Cleaner
Maintenance→Reset/Reboot
Settings→User Management→Operation Log
- “Admin” privilege users cannot create new users for login.

Note: By default, administrator accounts are not allowed to access backup menu, but this can be assigned to
them by navigating to Maintenance->User Management->Custom Privilege->Edit Admin Account and
giving the account the Backup privilege.
.

Figure 321: Assign Backup permission to "Admin" users

• 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, CDR of that extension, personal data and value-added features. For more
details; please refer to User Portal Guide.

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 - 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.

Figure 322: General User

• Custom Privilege

The Super Admin user can create users with different privileges. Following modules are available for privilege
customization.

- API Configuration
- Backup
- Callback
- Call Queue
- CDR Recording Files
- CDR Records
- CDR Statistics
- Conference
- Dial By Name
- DISA
- Emergency Calls
- Event List
- Extensions
- Outbound Routes
- Inbound Routes
- Fax/T.38
- Feature Codes
- IVR
- Paging/Intercom

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 - Parking Lot
- Pickup Groups
- PMS - Wakeup Service
- Ring Groups
- SCA
- Speed Dial
- System Status
- System Events
- Time Settings
- Video Conference
- Voicemail
- Voice Prompt
- Wakeup Service
- Zero Config

Figure 323: Create New Custom Privilege

System administrators can create custom privilege sets which can be assigned to created users.

Concurrent Multi-User Login

While UCM supports simultaneous user logins, if different users are modifying the same option or doing the
same operation, the following error message may appear:

Figure 324: Multiple User Operation Error Prompt

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Login Settings

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→Login settings page.

2. Enter the old password first.
3. Enter the new password and re-type the new password to confirm.
Note: If PBX Settings→General Settings→Enable Strong Password is toggled on, the minimum password
requirements are as follows:
• Must contain at least one number.
• Must contain at least one uppercase letter, lower case letter, OR special character.
4. Configure Email Address for account recovery purposes.
5. Click on "Save".
6. User will be automatically logged out.
7. Once the web page comes back to the login page again, enter the username "admin" and the new password
to login.

Figure 325: 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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 Email Address Configure an email address for account recovery purposes

Change Username Allows users to change Super Administrator username

Login Security

After the user logs in the UCM6200 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
UCM6200 web GUI→Maintenance→Login Settings→Login security page.

The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user does not make
any operation on Web GUI before 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 the username and password again to
log back in.
If set to 0, there will be no timeout period, and users can remain logged in for an indefinite period of time.

“Maximum number of login attempts” protects the UCM against brute force authentication attempts. If the
number of attempts from an IP address exceeds the configured value, it will be blacklisted, and the IP address
will be banned based on the configured User ban period. Default value is 5.

“User ban period” specifies the amount of time (in minutes) that an IP address is banned from attempting to
log into the UCM system. A value of 0 indicates a permanent ban. Default value is 5.

“Login Banned User List” shows the list of IP addresses banned from the UCM.

“Login Whitelist” Users can create a list of IP addresses that will not be banned even if they exceed the
maximum number of failed login attempts.

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 Figure 326: Login Timeout Settings

Operation Log

The Operation Log lists all activities done by all users that have accessed the UCM web portal. This page is
found under Maintenance->Operation Log.

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 Figure 327: Operation Logs

The operation log can be sorted and filtered for easy access. Click on the header of each column to sort based
on those columns. For example, clicking on "Date" will sort the logs according to operation date and time. Clicking
on "Date" again will reverse the order.

Table 144: Operation Log Column Header

Date The date and time when the operation is executed.

Username 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 to view the options and values configured by this operation.

Remark Allows users to add notes and remarks to each 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 328: Operation Logs Filter

In the figure above, an example that shows 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.

Users can either filter out specific operation logs or then click on the button to delete

the filtered results or click on the button to delete all operation logs. Note: Deleting operation
logs will generate an operation log entry detailing the action.

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.

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Upgrading Via Network

The UCM6200 can be upgraded via TFTP/HTTP/HTTPS by configuring the URL/IP Address for the
TFTP/HTTP/HTTPS server and selecting a download method. Configure a valid URL for TFTP, HTTP or HTTPS;
the server name can be FQDN or IP address.
Examples of valid URLs:
firmware.grandstream.com/BETA
The upgrading configuration can be accessed via Web GUI→Maintenance→Upgrade.

Figure 329: Network Upgrade

Table 145: 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 Username The username for the HTTP/HTTPS server.

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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 is appropriate to avoid immediate service interruption. After it
boots up, log in the Web GUI to check the firmware version.

Upgrading Via Local Upload

If there is no HTTP/TFTP server, users could also upload the firmware to the UCM6200 directly via Web GUI.
Please follow the steps below to upload firmware locally.

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 330: Local Upgrade

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 Figure 331: Upgrading Firmware Files

4. Wait until the upgrading process is successful and a window will be popped up in the Web GUI.

Figure 332: Reboot UCM6200

5. Click on "OK" to reboot the UCM6200 and check the firmware version after it boots up.

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Notes:
- Please do not interrupt or power cycle the UCM6200 during upgrading process.
- The firmware file name allows the use of the special characters besides the following restricted
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;

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.

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Backup/Restore

To create backups, navigate to Maintenance->Backup->Backup/Restore and click on the Backup button to

start the process.

Figure 333: Create New Backup

1. Choose the type(s) of files to be included in the backup.

2. Choose where to store the backup file: USB Disk, SD Card, Local or NAS.

Note: USB Disk or SD card options will show only if plugged; NAS server will show only if configured and
status is available. Refer to [PBX Settings/NAS].

3. Name the backup file.

4. Click on "Backup" to start backup.

Once the backup is done, it will appear in the List of Previous Configuration Backups. From this list, users

can download, restore, and delete backups. Users can download , restore , or delete it from the
UCM62xx internal storage or the external device.

Users can also upload UCM backups from their PC to the UCM by clicking on the Upload button.
Note: Uploaded backup files can only be 10MB or lower in file size.

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 Figure 334: Backup / Restore

The option allows UCM62xx UCM to perform periodic backups to USB disks, SD cards,
SFTP servers, or NAS servers. Users can configure the hour and frequency of the periodic backup.

Users can test connections with SFTP servers by clicking on the button.

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 Figure 335: Local Backup

Data Sync

Besides local backup, users can back up call recordings, voicemail, CDR, and fax daily to an SFTP server via
the Data Sync feature in the Maintenance->Backup->Data Sync page.

Special characters such as @ and periods (.) are supported, allowing users to use email addresses as SFTP
account usernames and to specify destination directories on the SFTP server. If a specified directory does not
exist on the SFTP server, the UCM will create it automatically.

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 Figure 336: Data Sync

Table 146: 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.

Before saving the configuration, users can click on the Test Connection button to verify connection status.
Clicking on the Synchronize All Data button will manually start a data sync.

Save the changes and all the backup logs will be listed on the web page.

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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 to overwrite the UCM's current settings with the backup's.

• 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.
• Users can restore backup files located on SD cards, USB disks, or NAS servers. also restore using the
backup file saved in SD card or USB device plugged into the UCM6200 or the backup available in his NAS
server connected to UCM.

Figure 337: 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.
• Uploaded backup file size cannot exceed 10MB.
---------------------------------------------------------------------------------------------------------------------------------------------------

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 type. There are two different reset types.

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• User Data
Voicemail, call recordings, voice prompts, Music on Hold files, CDR, and backup files will be erased.
• All
All files will be erased, and all settings will be reset to factory settings.

Figure 338: Reset and Reboot

Cleaner

The UCM can be configured to delete files, CDR, voicemail, and calls statistic reports on a regular basis. This
featurer can be accessed from the Maintenance->System Cleanup/Reset->Cleaner page.

The following screenshot show the settings and parameters to configure the cleaner feature on UCM6200.

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 Figure 339: Cleaner

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 Table 147: Cleaner Configuration

CDR Cleaner

Enable Cleaner Enable the CDR Cleaner function.

Clean Time Enter 0-23 to specify the hour of the day to clean up CDR.

Cleaning Conditions By Schedule: If the clean interval is 3, cleaning will be performed every 3
days to remove all records that were generated in the past 3 days.
• Keep Last X Records: If the max number of CDR has been reached,
CDR will be deleted starting with the oldest entry at the configured
cleaning time. (Note: The amount of records displayed on the page of
call queue statistics is not one-to-one with the actual amount of records
in the database.)

• Keep Last X Days: Delete all entries older than X days.

Clean Interval Enter 1-30 to specify the day of the month to clean up CDR when By
Schedule is selected as Cleaning Conditions.

Max Entries Set the maximum number of CDR entries to keep when Keep Last X
Records is selected as Cleaning Conditions.
Default is 50000. Valid range: 10000 – 100000.

Keep Last X Days Enter the number of days of call log entries to keep when Keep Last X days
is selected as Cleaning Conditions.
Default is 30. Valid range: 1 – 100.

Queue Statistics Report Cleaner

Enable Cleaner Enable scheduled queue log cleaning.

Clean Time Enter the hour of the day to start the cleaning. The valid range is 0-23.

Cleaning Conditions • By Schedule: If the clean interval is 3, cleaning will be performed every
3 days to remove all records that were generated 3 days ago.

• Keep Last X Records: If the max number of Queue Statistics Report

entries has been reached, Queue Statistics Report entries will be
deleted starting with the oldest entry at the configured cleaning time.
(Note: The amount of records displayed on the page of call queue
statistics is not one-to-one with the actual amount of records in the
database.)

• Keep Last X Days: Delete all entries older than X days.

Clean Interval Enter how often (in days) to clean queue logs when By Schedule is

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 selected as Cleaning Conditions. The valid range is 1-30.

Max Entries Set the maximum number of Queue Statistics Report entries to keep when
Keep Last X Records is selected as Cleaning Conditions.
Default is 50000. Valid range: 10000 – 100000.

Keep Last X Days Enter the number of days of Queue Statistics Report entries to keep when
Keep Last X days is selected as Cleaning Conditions.
Default is 30. Valid range: 1 – 100.

Conference Call Statistics Report Cleaner

Enable Cleaner Enable scheduled conference log cleaning.

Clean time Enter the hour of the day to start the cleaning. The valid range is 0-23.

Cleaning Conditions • By Schedule: If the clean interval is 3, cleaning will be performed every
3 days to remove all records that were generated 3 days ago.

• Keep Last X Records: If the max number of Conference Call Statistics

Report has been reached, Conference Call Statistics Report will be
deleted starting with the oldest entry at the configured cleaning time.
(Note: The amount of records displayed on the page of call queue
statistics is not one-to-one with the actual amount of records in the
database.)

• Keep Last X Days: Delete all entries older than X days.

Clean Interval Enter how often (in days) to clean queue logs when By Schedule is
selected as Cleaning Conditions. The valid range is 1-30.

Max Entries Set the maximum number of CDR Conference Call Statistics Report entries
to keep when Keep Last X Records is selected as Cleaning Conditions.
Default is 50000. Valid range: 10000 – 100000.

Keep Last X Days Enter the number of days of Conference Call Statistics Report entries to
keep when Keep Last X days is selected as Cleaning Conditions.
Default is 30. Valid range: 1 – 100.

File Cleaner

Enable Cleaner Enables file cleaning

Clean Files in External If enabled the files in external device (USB/SD card) will be automatically
Device cleaned up as configured.

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 Choose Cleaner File Select the files for system automatic clean.
• Basic Call Recording Files.
• Conference Recording Files.
• Call Queue Recording Files.
• Voicemail Files.
• Emergency Calls Recording Files.
• Fax.
• Backup Files.
• SCA Recording Files.

Clean Time Enter the hour of the day to start the cleaning. The valid range is 0-23.

Cleaning Conditions • By Schedule: If the clean interval is 3, cleaning will be performed

every 3 days to delete all files.

• By Threshold: Check at the configured cleaning time every day to

see if the storage threshold has been exceeded and perform cleaning
of all files if it has.

• Keep Last X Days: Delete all files older than X days.

File Clean Interval Enter 1-30 to specify the day of the month to clean up the files.

File Clean Threshold Enter the internal storage disk usage threshold (in percent). Once this
threshold is exceeded, the file cleanup will proceed as scheduled. Valid
range is 0-99.

Keep Last X Days Automatically delete all recordings older than this x days when the threshold
is reached. If not set, all data is cleared. Valid range: 1 – 100.

Cleaner Log

Cleaner Log Clicking on the Clean button will clear the cleaner log.

All the cleaner logs will be listed on the bottom of the page.

USB Disk/SD Card File Management

From this page, users can view, download, and delete files on USB disks and SD cards.

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 Figure 340: USB/SD Card Files Cleanup
Table 148: 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.

System Recovery

In the scenario where the UCM cannot be accessed via web or SSH, users can try to recover the system by
resetting the system or upgrading system firmware via the UCM's recovery mode.

To access recovery mode, follow these steps:

1. Disconnect the power cable from the UCM and keep the network cable connected.
2. Press and hold the reset button located on the back of the UCM.

3. Reconnect the power cable while holding the reset button.

4. Keep holding until a click sound is heard. Release the reset button.

The UCM's LCD should now display "Recovery Mode" and an IP address.

Once at this stage, the administrator can access the recovery mode web portal by typing in either the IP0 address
(typically WAN) or IP1 address (typically LAN) into a browser address bar. The following page should appear:

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 Figure 341: UCM6202 Recovery Web Page

Enter the super admin's login credentials and click on the Login button.

Figure 342: Recovery Mode

From here, the user can either upload a firmware file, factory reset or just reboot the device.

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Syslog

On the Maintenance->Syslog page, users can configure the UCM to send syslog to a remote server, to do so,
please refer to below options:

Syslog server The URL/IP address for the syslog server.

Write Syslog Into Cache If enabled, syslog will be sent to and cached in system memory until it is
written to internal storage at the specified time interval. Enabling this
helps to improve the lifespan of the UCM's flash storage
Time Interval(seconds) Configures the period of time that must pass before writing the cached
syslog to internal storage. Setting the time interval too high may result in
lost syslog data due to the low saving frequency.

Figure 343: Syslog Server

By default, ERROR logs are enabled for all PBX modules. Additional syslog levels are WARN, NOTICE, DEBUG,
and VERBOSE.

• 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:
Syslogs are for debugging and troubleshooting. It is not recommended to enable syslog for all modules and
levels, as it can cause excessive resource usage and network load.
UCM reserves a 50MB cache for syslog. Once the syslog has reached 50MB in file size, 2MB of syslog will be
deleted, starting from the oldest syslogs. This ensures that the system can continue logging.
----------------------------------------------------------------------------------------------------------------------------- ----------------------

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Network Troubleshooting

The Network Troubleshooting page under Maintenance allows users to start network captures and
ping/traceroute remote hosts.

Ethernet Capture

Captured traces can be downloaded for analysis.

Figure 344: Ethernet Capture

Table 149: 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.
Capture Filter Enter the filter to obtain the specific types of traffic, such as (host,
src, dst, net, proto…).

Save to External Storage Check this box to activate storage of the capture either on the
USB or SD Card.

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 .tgz format, and users will need to use a compression tool such as WinRAR or 7-zip to

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unzip the .pcap file.
Users could specify the capture filter as used in general network traffic capture tool (host, src, dst, net, protocol,
port, port range) before starting to capture the trace.

Note: Capture files saved on external devices will now have “capture” prepended to file names.

IP Ping

Enter the hostname or IP address and click on the Start button. The ping process can be viewed from the
Output Result window at the bottom of the page.

Figure 345: Ping

Traceroute

Enter the hostname or IP address and click on the Start button. The ping process can be viewed from the
Output Result window at the bottom of the page.

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 Figure 346: Traceroute

Signaling Troubleshooting

Analog Record Trace

• Analog Record Trace

Analog record traces are used to troubleshoot analog trunk issues (e.g., CID detection for incoming calls from
an analog trunk.

To capture analog record traces:

1. Select FXO or FXS for "Record Ports". If the issue happens on FXO 1, select FXO port 1 to record the
trace.

2. Select "Record Direction".

3. Select "Record File Mode" to separate the record per direction or mix.

4. Click on "Start".

5. Make a call via the analog port that has the issue.

6. Once done, click on "Stop".

7. Click on "Download" to download the analog record trace.

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 Figure 347: Troubleshooting Analog Trunks

• key Dial-up FXO

Users can configure a PSTN number in the External Number field to send test calls to and troubleshoot
issues related to the used analog trunk. To use this:

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 348: A Key Dial-up FXO

Note: To use Key dial-up FXO troubleshooting, the outbound route for the analog trunk must have permission
set to "Internal" and must have the highest priority for the external number used for troubleshooting. After
capturing the trace, users can download it for basic analysis. If further assistance is necessary, please contact
Grandstream Technical Support using the following link:
http://www.grandstream.com/index.php/support

• SVIP Record Trace

Users can capture an SVIP record trace for specific FXO/FXS ports.

Figure 349: SVIP Record Trace

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Service Check

Service Check will cause the UCM to periodically send SIP OPTIONS to the Asterisk server to check on
functionality.

Check Cycle is the frequency (in seconds) at which to check the Asterisk server for issues.

Check Times is the maximum number of failed checks that can occur before the UCM is restart, and a coredump
file is generated.

Generated coredump files can be sent to Grandstream Technical Support for analysis and troubleshooting.

Figure 350: Service Check

Network Status

Users can navigate to System Status->Network Status to view the UCM's active network connections. If
users are experiencing connection issues to specific services, they can view this page for troubleshooting.

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 Figure 351: 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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