Grandstream Networks, Inc.
UCM6510 IP PBX
User Manual
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©2016 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:
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Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc.
in the United States, Europe and other countries.
CAUTION
Changes or modifications to this product not expressly approved by Grandstream, or operation of this
product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
WARNING
Please do not use a different power adaptor with your devices as it may cause damage to the products
and void the manufacturer warranty.
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GNU GPL INFORMATION
UCM6510 firmware contains third-party software licensed under the GNU General Public License (GPL).
Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public
License (GPL) for the exact terms and conditions of the license.
Grandstream GNU GPL related source code can be downloaded from Grandstream web site from:
http://www.grandstream.com/support/faq/gnu-general-public-license/gnu-gpl-information-download
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Table of Content
DOCUMENT PURPOSE........................................................................... 20
CHANGE LOG ......................................................................................... 21
Firmware Version 1.0.11.27 ................................................................................................................. 21
Firmware Version 1.0.10.44................................................................................................................. 21
Firmware Version 1.0.10.39................................................................................................................. 22
Firmware Version 1.0.2.7 ..................................................................................................................... 22
Firmware Version 1.0.2.5 ..................................................................................................................... 23
Firmware Version 1.0.1.12................................................................................................................... 24
Firmware Version 1.0.0.25................................................................................................................... 25
WELCOME ............................................................................................... 26
PRODUCT OVERVIEW ............................................................................ 27
Feature Highlights ............................................................................................................................... 27
Technical Specifications ...................................................................................................................... 27
INSTALLATION ........................................................................................ 30
Equipment Packaging.......................................................................................................................... 30
Connect your UCM6510 ...................................................................................................................... 30
Connect The UCM6510 ................................................................................................................ 30
Safety Compliances ............................................................................................................................. 32
Warranty .............................................................................................................................................. 32
GETTING STARTED ................................................................................ 33
Use The LCD Menu ............................................................................................................................. 33
Use The LED Indicators ...................................................................................................................... 35
Use The Web GUI ............................................................................................................................... 36
Access Web GUI .......................................................................................................................... 36
Setup Wizard ................................................................................................................................ 37
Web GUI Configurations ............................................................................................................... 38
Web GUI Languages .................................................................................................................... 38
Save And Apply Changes ............................................................................................................. 39
Make Your First Call ............................................................................................................................ 39
SYSTEM SETTINGS ................................................................................ 41
User Management ............................................................................................................................... 41
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User Privileges .............................................................................................................................. 41
Create New Web UI User ............................................................................................................. 42
User Portal .................................................................................................................................... 44
Concurrent Multi-User Login ......................................................................................................... 45
Operation Log ............................................................................................................................... 45
Change Password ............................................................................................................................... 47
Change Binding Email .................................................................................................................. 48
Network Settings ................................................................................................................................. 49
Basic Settings ............................................................................................................................... 49
DHCP Client List ........................................................................................................................... 54
802.1X .......................................................................................................................................... 55
Static Routes ................................................................................................................................ 57
Port Forwarding ............................................................................................................................ 59
DDNS Settings .............................................................................................................................. 61
Firewall ................................................................................................................................................ 62
Static Defense .............................................................................................................................. 63
Dynamic Defense ......................................................................................................................... 65
Fail2ban ........................................................................................................................................ 66
LDAP Server ........................................................................................................................................ 67
LDAP Server Configurations......................................................................................................... 68
LDAP Phonebook ......................................................................................................................... 69
LDAP Client Configurations .......................................................................................................... 72
HTTP Server ........................................................................................................................................ 74
Email Settings ...................................................................................................................................... 75
Email Settings ............................................................................................................................... 75
Email Templates ........................................................................................................................... 77
Time Settings ....................................................................................................................................... 78
Auto Time Updating ...................................................................................................................... 78
Set Time Manually ........................................................................................................................ 80
Office Time ................................................................................................................................... 80
Holiday .......................................................................................................................................... 82
NTP Server .......................................................................................................................................... 84
Recordings Storage ............................................................................................................................. 85
Login Settings ...................................................................................................................................... 87
Google Service Settings Support ........................................................................................................ 88
PROVISIONING ....................................................................................... 91
Overview .............................................................................................................................................. 91
Configuration Architecture for End Point Device ................................................................................. 91
Auto Provisioning Settings................................................................................................................... 92
Discovery ............................................................................................................................................. 95
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Global Configuration ............................................................................................................................ 96
Global Policy ................................................................................................................................. 96
Global Templates ........................................................................................................................ 103
Model Configuration .......................................................................................................................... 106
Model Templates ........................................................................................................................ 106
Model Update ............................................................................................................................. 108
Device Configuration ......................................................................................................................... 109
Create New Device ..................................................................................................................... 109
Manage Devices ......................................................................................................................... 110
Sample Application ............................................................................................................................ 116
EXTENSIONS......................................................................................... 121
Create New User ............................................................................................................................... 121
Create New SIP Extension ......................................................................................................... 121
Create New IAX Extension ......................................................................................................... 127
Create New FXS Extension ........................................................................................................ 131
Batch Add Extensions ........................................................................................................................ 136
Batch Add SIP Extensions .......................................................................................................... 136
Batch Add IAX Extensions .......................................................................................................... 139
Search and Edit Extension ................................................................................................................ 141
Export Extensions .............................................................................................................................. 143
Import Extensions .............................................................................................................................. 143
Email to User ..................................................................................................................................... 144
Multiple Registrations Per Extension ................................................................................................. 146
SMS Message Support ...................................................................................................................... 147
ANALOG TRUNKS ................................................................................ 149
Analog Trunks Configuration ............................................................................................................. 149
PSTN Detection ................................................................................................................................. 152
Analog Hardware Configuration ........................................................................................................ 155
DIGITAL TRUNKS .................................................................................. 158
Digital Hardware Configuration ......................................................................................................... 158
Digital Trunk Configuration ................................................................................................................ 168
Direct Outward Dialing (DOD) via Digital Trunks .............................................................................. 169
Digital Trunk Troubleshooting ............................................................................................................ 169
DATA TRUNK ......................................................................................... 171
VOIP TRUNKS ....................................................................................... 173
VoIP Trunk Configuration ................................................................................................................... 173
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Direct Outward Dialing (DOD) via VoIP Trunks ................................................................................. 183
SLA STATION ........................................................................................ 185
Create/Edit SLA Station ..................................................................................................................... 185
Sample Configuration ........................................................................................................................ 186
CALL ROUTES ...................................................................................... 188
Outbound Routes .............................................................................................................................. 188
Outbound Routes........................................................................................................................ 188
Country Codes ............................................................................................................................ 190
Inbound Routes ................................................................................................................................. 191
Inbound Rule Configurations ...................................................................................................... 191
Inbound Route: Prepend Example ............................................................................................. 194
Inbound Route: Multiple Mode .................................................................................................... 195
Fax Intelligent Route ................................................................................................................... 195
Fax with Two Media .................................................................................................................... 196
Blacklist Configurations .............................................................................................................. 196
CONFERENCE BRIDGE ........................................................................ 198
Conference Bridge Configurations ............................................................................................. 198
Join a Conference Call ............................................................................................................... 199
Invite Other Parties to Join Conference...................................................................................... 200
During The Conference .............................................................................................................. 200
Record Conference..................................................................................................................... 202
CONFERENCE SCHEDULE .................................................................. 203
Conference Schedule Configuration ................................................................................................. 203
IVR ......................................................................................................... 207
Configure IVR .................................................................................................................................... 207
Create IVR Prompt ............................................................................................................................ 209
Record New IVR Prompt ............................................................................................................ 209
Upload IVR Prompt ..................................................................................................................... 210
LANGUAGE SETTINGS FOR VOICE PROMPT .................................... 211
Download and Install Voice Prompt Package.................................................................................... 211
Customize Specific Prompt ............................................................................................................... 213
VOICEMAIL ............................................................................................ 215
Configure Voicemail ........................................................................................................................... 215
Access Voicemail ............................................................................................................................... 216
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Voicemail Email Settings ................................................................................................................... 217
Configure Voicemail Group................................................................................................................ 219
RING GROUP......................................................................................... 220
Configure Ring Group ........................................................................................................................ 220
Remote Extension in Ring Group ...................................................................................................... 222
PAGING AND INTERCOM GROUP ....................................................... 224
Configure Paging/Intercom Group ..................................................................................................... 224
CALL QUEUE ........................................................................................ 226
Configure Call Queue ........................................................................................................................ 226
EXTENSION GROUPS........................................................................... 230
Configure Extension Groups ............................................................................................................. 230
Use Extension Groups ....................................................................................................................... 230
PICKUP GROUPS .................................................................................. 232
Configure Pickup Groups .................................................................................................................. 232
Configure Pickup Feature Code ........................................................................................................ 232
MUSIC ON HOLD ................................................................................... 234
FAX/T.38 ................................................................................................. 236
Configure Fax/T.38 ............................................................................................................................ 236
Sample Configuration to Receive Fax from PSTN Line .................................................................... 237
Sample Configuration for Fax-To-Email ............................................................................................ 240
ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) ............ 242
FOLLOW ME .......................................................................................... 243
ONE-KEY DIAL ...................................................................................... 246
DISA ....................................................................................................... 248
CALLBACK FEATURE .......................................................................... 250
BLF AND EVENT LIST........................................................................... 252
BLF .................................................................................................................................................... 252
Event List ........................................................................................................................................... 252
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DIAL BY NAME ...................................................................................... 255
Dial By Name Configuration .............................................................................................................. 255
ACTIVE CALLS AND MONITOR ........................................................... 259
Active Calls Status ............................................................................................................................. 259
Hang Up Active Calls ......................................................................................................................... 261
Call Monitor ....................................................................................................................................... 261
CALL FEATURES .................................................................................. 264
Feature Codes ................................................................................................................................... 264
Call Recording ................................................................................................................................... 268
Call Park ............................................................................................................................................ 269
Park a Call .................................................................................................................................. 269
Retrieve The Parked Call ........................................................................................................... 269
Enable Spy ........................................................................................................................................ 269
INTERNAL OPTIONS ............................................................................. 271
Internal Options/General ................................................................................................................... 271
Internal Options/Jitter Buffer .............................................................................................................. 273
Internal Options/RTP Settings ........................................................................................................... 274
Internal Options/Payload ................................................................................................................... 274
Internal Options/PIN Groups ............................................................................................................. 275
IAX SETTINGS ....................................................................................... 278
IAX Settings/General ......................................................................................................................... 278
IAX Settings/Registration ................................................................................................................... 278
IAX Settings/Static Defense .............................................................................................................. 279
SIP SETTINGS ....................................................................................... 280
SIP Settings/General ......................................................................................................................... 280
SIP Settings/Misc .............................................................................................................................. 280
SIP Settings/Session Timer ............................................................................................................... 281
SIP Settings/TCP and TLS ................................................................................................................ 281
SIP Settings/NAT ............................................................................................................................... 283
SIP Settings/TOS............................................................................................................................... 283
VALUE-ADDED FEATURES .................................................................. 285
Fax Sending ...................................................................................................................................... 285
Announcements Center ..................................................................................................................... 286
Announcements Center Setting .................................................................................................. 286
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Group Setting .............................................................................................................................. 287
PMS (Property Management System) .................................................. 289
Basic Settings .................................................................................................................................... 289
Room Status ...................................................................................................................................... 289
Wake Up Service ............................................................................................................................... 291
STATUS AND REPORTING ................................................................... 293
PBX Status ........................................................................................................................................ 293
Trunks ......................................................................................................................................... 293
Extensions .................................................................................................................................. 294
Queues ....................................................................................................................................... 296
Conference Rooms ..................................................................................................................... 297
Interfaces Status ......................................................................................................................... 297
Digital Channels Status .............................................................................................................. 299
Parking Lot .................................................................................................................................. 300
System Status ................................................................................................................................... 301
General ....................................................................................................................................... 301
Network ....................................................................................................................................... 302
Storage Usage ............................................................................................................................ 302
Resource Usage ......................................................................................................................... 303
System Events ................................................................................................................................... 304
Alert Events List .......................................................................................................................... 304
Alert Log ..................................................................................................................................... 307
Alert Contact ............................................................................................................................... 309
CDR ................................................................................................................................................... 309
CDR Improvement ...................................................................................................................... 313
Downloaded CDR File ................................................................................................................ 313
Statistics ..................................................................................................................................... 315
Recording Files ........................................................................................................................... 315
API Configuration Files ............................................................................................................... 316
UPGRADING AND MAINTENANCE ...................................................... 322
Upgrading .......................................................................................................................................... 322
Upgrading via Network ............................................................................................................... 322
Upgrading via Local Upload ....................................................................................................... 323
No Local Firmware Servers ........................................................................................................ 325
Backup ............................................................................................................................................... 325
Backup/Restore .......................................................................................................................... 325
Data Sync ................................................................................................................................... 327
Restore Configuration from Backup File .................................................................................... 328
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Cleaner .............................................................................................................................................. 329
Reset and Reboot .............................................................................................................................. 330
Syslog ................................................................................................................................................ 331
Troubleshooting ................................................................................................................................. 332
Ethernet Capture ........................................................................................................................ 332
IP Ping ........................................................................................................................................ 332
Traceroute ................................................................................................................................... 333
PRI/SS7/MFC/R2 Signaling Trace .............................................................................................. 334
Analog Record Trace .................................................................................................................. 334
E&M Immediate Record Trace.................................................................................................... 335
Service Check ............................................................................................................................. 335
Network Status ........................................................................................................................... 335
Remote Access .................................................................................................................................. 336
SSH Access ................................................................................................................................ 336
EXPERIENCING THE UCM6510 SERIES IP PBX ................................. 338
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Table of Tables
Table 1: Technical Specifications ................................................................................................................ 27
Table 2: UCM6510 Equipment Packaging .................................................................................................. 30
Table 3: LCD Menu Options ........................................................................................................................ 34
Table 4: UCM6510 LED INDICATORS ....................................................................................................... 35
Table 5: User Management – Create New User ......................................................................................... 43
Table 6: Operation Log Column Header ..................................................................................................... 46
Table 7: Change Binding Email option ........................................................................................................ 49
Table 8: UCM6510 Network Settings->Basic Settings ................................................................................ 49
Table 9: UCM6510 Network Settings->802.1X ........................................................................................... 56
Table 10: UCM6510 Network Settings->Static Routes ............................................................................... 57
Table 11: UCM6510 Network Settings->Port Forwarding ........................................................................... 59
Table 12: UCM6510 Firewall->Static Defense->Current Service................................................................ 63
Table 13: Typical Firewall Settings .............................................................................................................. 63
Table 14: Firewall Rule Settings .................................................................................................................. 64
Table 15: UCM6510 Firewall Dynamic Defense ......................................................................................... 65
Table 16: Fail2Ban Settings ........................................................................................................................ 66
Table 17: HTTP Server Settings .................................................................................................................. 75
Table 18: Email Settings .............................................................................................................................. 75
Table 19: Auto Time Updating ..................................................................................................................... 79
Table 20: Create New Office Time .............................................................................................................. 81
Table 21: Create New Holiday..................................................................................................................... 83
Table 22: Auto Provision Settings ............................................................................................................... 94
Table 23: Global Policy Parameters – Localization ..................................................................................... 97
Table 24: Global Policy Parameters – Phone Settings ............................................................................... 98
Table 25: Global Policy Parameters – Contact List ..................................................................................... 99
Table 26: Global Policy Parameters – Maintenance ................................................................................. 100
Table 27: Global Policy Parameters – Network Settings .......................................................................... 102
Table 28: Global Policy Parameters – Customization ............................................................................... 103
Table 29: Create New Template ................................................................................................................ 104
Table 30: Create New Model Template ..................................................................................................... 106
Table 31: SIP Extension Configuration Parameters – Basic Settings ....................................................... 122
Table 32: SIP Extension Configuration Parameters – Media .................................................................... 123
Table 33: SIP Extension Configuration Parameters – Features ............................................................... 124
Table 34: SIP Extension Configuration Parameters – Specific Time ........................................................ 127
Table 35: IAX Extension Configuration Parameters – Basic Settings ....................................................... 127
Table 36: IAX Extension Configuration Parameters – Media .................................................................... 128
Table 37: IAX Extension Configuration Parameters – Features ............................................................... 129
Table 38: IAX Extension Configuration Parameters – Specific Time ........................................................ 131
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Table 39: FXS Extension Configuration Parameters – Basic Settings ..................................................... 132
Table 40: FXS Extension Configuration Parameters – Media................................................................... 133
Table 41: FXS Extension Configuration Parameters – Features .............................................................. 134
Table 42: FXS Extension Configuration Parameters – Specific Time ....................................................... 136
Table 43: Batch Add SIP Extension Parameters ....................................................................................... 137
Table 44: Batch Add IAX Extension Parameters ....................................................................................... 139
Table 45: Analog Trunk Configuration Parameters ................................................................................... 149
Table 46: PSTN Detection for Analog Trunk ............................................................................................. 154
Table 47: Analog Hardware Configuration Parameters ............................................................................. 156
Table 48: Digital Hardware Configuration Parameters: E1 – PRI_NET/PRI_CPE ................................... 159
Table 49: Digital Hardware Configuration Parameters: E1 - SS7 ............................................................. 161
Table 50: Digital Hardware Configuration Parameters: E1 - MFC/R2 ...................................................... 162
Table 51: Digital Hardware Configuration Parameters: T1/J1 - PRI_NET/PRI_CPE ................................ 164
Table 52: Digital Hardware Configuration Parameters: T1/J1 - SS7......................................................... 166
Table 53: Digital Hardware Configuration Parameters: T1-E&M Immediate/E&M Wink ........................... 167
Table 54: Digital Trunk Configuration Parameters .................................................................................... 168
Table 55: Data Trunk Configuration Parameters ....................................................................................... 172
Table 56: Create New SIP Trunk ............................................................................................................... 173
Table 57: SIP Register Trunk Configuration Parameters .......................................................................... 174
Table 58: SIP Peer Trunk Configuration Parameters ................................................................................ 178
Table 59: Create New IAX Trunk ............................................................................................................... 180
Table 60: IAX Register Trunk Configuration Parameters .......................................................................... 181
Table 61: IAX Peer Trunk Configuration Parameters ................................................................................ 182
Table 62: SLA Station Configuration Parameters ..................................................................................... 185
Table 63: Outbound Route Configuration Parameters .............................................................................. 188
Table 64: Inbound Rule Configuration Parameters ................................................................................... 191
Table 65: Conference Bridge Configuration Parameters .......................................................................... 198
Table 66: Conference Caller IVR Menu .................................................................................................... 201
Table 67: Conference Schedule Parameters ............................................................................................ 203
Table 68: IVR Configuration Parameters .................................................................................................. 207
Table 69: Voicemail Settings ..................................................................................................................... 215
Table 70: Voicemail IVR Menu .................................................................................................................. 216
Table 71: Voicemail Email Settings ........................................................................................................... 218
Table 72: Voicemail Group Settings .......................................................................................................... 219
Table 73: Ring Group Parameters ............................................................................................................ 220
Table 74: Paging/Intercom Group Configuration Parameters ................................................................... 224
Table 75: Call Queue Configuration Parameters ...................................................................................... 226
Table 76: FAX/T.38 Settings ...................................................................................................................... 236
Table 77: Follow Me Settings .................................................................................................................... 244
Table 78: Follow Me Options ..................................................................................................................... 245
Table 79: DISA Settings ............................................................................................................................ 248
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Table 80: Callback Configuration Parameters........................................................................................... 250
Table 81: Event List Settings ..................................................................................................................... 253
Table 82: UCM6510 Feature Codes ......................................................................................................... 264
Table 83: Internal Options/General ........................................................................................................... 271
Table 84: Internal Options/Jitter Buffer ...................................................................................................... 273
Table 85: Internal Options/RTP Settings ................................................................................................... 274
Table 86: Internal Options/Payload ........................................................................................................... 274
Table 87: PIN Group .................................................................................................................................. 275
Table 88: IAX Settings/General ................................................................................................................. 278
Table 89: IAX Settings/Registration .......................................................................................................... 278
Table 90: IAX Settings/Static Defense ...................................................................................................... 279
Table 91: SIP Settings/General ................................................................................................................. 280
Table 92: SIP Settings/Misc ...................................................................................................................... 280
Table 93: SIP Settings/Session Timer ....................................................................................................... 281
Table 94: SIP Settings/TCP and TLS ........................................................................................................ 281
Table 95: SIP Settings/NAT ....................................................................................................................... 283
Table 96: SIP Settings/ToS........................................................................................................................ 283
Table 97: Announcements Center Setting ................................................................................................. 286
Table 98: Group Setting ............................................................................................................................ 287
Table 99: PMS Basic Settings ................................................................................................................... 289
Table 100: PMS Wake up Service ............................................................................................................. 292
Table 101: Trunk Status ............................................................................................................................ 293
Table 102: Extension Status...................................................................................................................... 295
Table 103: Agent Status ............................................................................................................................ 296
Table 104: Interface Status Indicators ....................................................................................................... 297
Table 105: Digital Channel Status Indicators ............................................................................................ 300
Table 106: Parking Lot Status ................................................................................................................... 301
Table 107: System Status->General ......................................................................................................... 302
Table 108: System Status->Network ......................................................................................................... 302
Table 109: CDR Filter Criteria ................................................................................................................... 309
Table 110: CDR Statistics Filter Criteria .................................................................................................... 315
Table 111: CDR API Configuration Files.................................................................................................... 316
Table 112: CDR API URI Parameters........................................................................................................ 317
Table 113: Network Upgrade Configuration .............................................................................................. 323
Table 114: Data Sync Configuration .......................................................................................................... 328
Table 115: Cleaner Configuration .............................................................................................................. 330
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Table of Figures
Figure 1: UCM6510 Front View................................................................................................................... 30
Figure 2: UCM6510 Back View ................................................................................................................... 30
Figure 3: UCM6510 T1/E1/J1 Crossover Cable Pin-out ............................................................................. 31
Figure 4: UCM6510 web GUI Login Page .................................................................................................. 36
Figure 5: UCM6510 Setup Wizard .............................................................................................................. 37
Figure 6: UCM6510 web GUI Language ..................................................................................................... 39
Figure 7: UCM6510 web GUI: Apply Changes ........................................................................................... 39
Figure 8: User Management Page Display ................................................................................................. 41
Figure 9: Create New User ......................................................................................................................... 43
Figure 10: User Management – New Users ................................................................................................ 43
Figure 11: Edit User Information by Super Admin ....................................................................................... 44
Figure 12: User Portal Login ....................................................................................................................... 44
Figure 13: User Portal Layout ..................................................................................................................... 45
Figure 14: Multiple User Operation Error Prompt ....................................................................................... 45
Figure 15: Operation Logs .......................................................................................................................... 46
Figure 16: Operation Logs Filter ................................................................................................................. 47
Figure 17: Change Password...................................................................................................................... 48
Figure 18: Change Binding Email ............................................................................................................... 48
Figure 19: UCM6510 Network Interface Method: Route ............................................................................. 52
Figure 20: UCM6510 Network Interface Method: Switch ............................................................................ 53
Figure 21: UCM6510 Network Interface Method: Dual ............................................................................... 54
Figure 22: DHCP Client List ........................................................................................................................ 54
Figure 23: Add MAC Address Bind ............................................................................................................. 55
Figure 24: Batch Add MAC Address Bind ................................................................................................... 55
Figure 25: UCM6510 Using 802.1X as Client ............................................................................................. 56
Figure 26: UCM6510 Using 802.1X EAP-MD5 ........................................................................................... 56
Figure 27: UCM6510 Static Route Sample ................................................................................................. 58
Figure 28: UCM6510 Static Route Configuration........................................................................................ 59
Figure 29: UCM6510 Port Forwarding Configuration ................................................................................. 60
Figure 30: GXP2160 Web Access Using UCM6510 Port Forwarding ........................................................ 61
Figure 31: Register Domain Name on noip.com ......................................................................................... 61
Figure 32: UCM6510 DDNS Setting ........................................................................................................... 62
Figure 33: Using Domain Name to Connect to UCM6510 .......................................................................... 62
Figure 34: Create New Firewall Rule .......................................................................................................... 64
Figure 35: Configure Dynamic Defense ...................................................................................................... 66
Figure 36: LDAP Server Configurations ...................................................................................................... 68
Figure 37: Default LDAP Phonebook DN .................................................................................................... 68
Figure 38: Default LDAP Phonebook Attributes .......................................................................................... 69
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Figure 39: Add LDAP Phonebook ............................................................................................................... 69
Figure 40: Edit LDAP Phonebook ............................................................................................................... 70
Figure 41: Import Phonebook...................................................................................................................... 70
Figure 42: Phonebook CSV File Format ..................................................................................................... 71
Figure 43: LDAP Phonebook After Import ................................................................................................... 71
Figure 44: Export Selected LDAP Phonebook ............................................................................................ 72
Figure 45: LDAP Client Configurations ....................................................................................................... 73
Figure 46: GXP2200 LDAP Phonebook Configuration ............................................................................... 74
Figure 47: UCM6510 Email Settings ........................................................................................................... 76
Figure 48: UCM6510 Email Settings: Send Test Email............................................................................... 77
Figure 49: Email Templates ......................................................................................................................... 77
Figure 50: Conference Schedule Template ................................................................................................. 78
Figure 51: Set Time Manually ..................................................................................................................... 80
Figure 52: Create New Office Time ............................................................................................................. 81
Figure 53: Settings->Time Settings->Office Time ....................................................................................... 82
Figure 54: Create New Holiday ................................................................................................................... 82
Figure 55: Settings->Time Settings->Holiday ............................................................................................. 83
Figure 56: Settings->Recordings Storage ................................................................................................... 85
Figure 57: Recordings Storage Prompt Information ................................................................................... 86
Figure 58: Recording Storage Category ..................................................................................................... 86
Figure 59: Login Timeout Settings .............................................................................................................. 87
Figure 60: Google Service Settings: OAuth2.0 Authentication ................................................................... 88
Figure 61: Google Service: New Project ..................................................................................................... 89
Figure 62: Google Service: Create new credential ..................................................................................... 89
Figure 63: Google Service: OAuth2.0 login ................................................................................................ 90
Figure 64: Zero Config Configuration Architecture for End Point Device ................................................... 92
Figure 65: UCM6510 Zero Config ............................................................................................................... 93
Figure 66: Auto Provision Settings .............................................................................................................. 94
Figure 67: Auto Discover ............................................................................................................................. 96
Figure 68: Discovered Devices ................................................................................................................... 96
Figure 69: Global Policy Categories ........................................................................................................... 97
Figure 70: Edit Global Template ................................................................................................................ 105
Figure 71: Edit Model Template ................................................................................................................ 107
Figure 72: Template Management ............................................................................................................ 108
Figure 73: Upload Model Template Manually ............................................................................................ 108
Figure 74: Create New Device .................................................................................................................. 109
Figure 75: Manage Devices ...................................................................................................................... 110
Figure 76: Edit Device ............................................................................................................................... 110
Figure 77: Edit Customize Device Settings ............................................................................................... 112
Figure 78: Add P Value in Customize Device Settings ............................................................................. 113
Figure 79: Modify Selected Devices–Same Model ................................................................................... 114
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Figure 80: Modify Selected Devices—Different Models............................................................................ 115
Figure 81: Device List in Zero Config ........................................................................................................ 116
Figure 82: Zero Config Sample – Global Policy ........................................................................................ 117
Figure 83: Zero Config Sample – Device Preview 1 ................................................................................. 118
Figure 84: Zero Config Sample – Device Preview 2 ................................................................................. 119
Figure 85: Zero Config Sample – Device Preview 3 ................................................................................. 120
Figure 86: Create New Device .................................................................................................................. 121
Figure 87: Manage Extensions ................................................................................................................. 142
Figure 88: Export Extensions .................................................................................................................... 143
Figure 89: Export Extensions .................................................................................................................... 143
Figure 90: Email To User: Prompt Information .......................................................................................... 144
Figure 91: Email To User: Account Registration Information and QR Code ............................................. 145
Figure 92: Email To User: LDAP Client Information and QR Code ........................................................... 146
Figure 93: Multiple Registrations per Extension ....................................................................................... 146
Figure 94: Extension - Concurrent Registration ........................................................................................ 147
Figure 95: SMS Message Support ............................................................................................................ 147
Figure 96: UCM6510 FXO Tone Settings ................................................................................................. 152
Figure 97: UCM6510 PSTN Detection ...................................................................................................... 153
Figure 98: UCM6510 PSTN Detection: Auto Detect ................................................................................. 153
Figure 99: UCM6510 PSTN Detection: Semi-Auto Detect ....................................................................... 154
Figure 100: FXS Ports Signaling Preference ............................................................................................ 155
Figure 101: FXO Ports ACIM Settings ...................................................................................................... 156
Figure 102: Digital Hardware Configuration .............................................................................................. 158
Figure 103: Troubleshooting Digital Trunks .............................................................................................. 170
Figure 104: Data Trunk Web Page............................................................................................................ 171
Figure 105: Data Trunk Configuration ....................................................................................................... 171
Figure 106: DOD extension selection ....................................................................................................... 184
Figure 107: Edit DOD ................................................................................................................................ 184
Figure 108: SLA Station ............................................................................................................................ 185
Figure 109: Enable SLA Mode for Analog Trunk ....................................................................................... 186
Figure 110: Analog Trunk with SLA Mode Enabled ................................................................................... 186
Figure 111: SLA Example - SLA Station .................................................................................................... 187
Figure 112: SLA Example - MPK Configuration ........................................................................................ 187
Figure 113: Country Codes ....................................................................................................................... 191
Figure 114: Inbound Route feature: Prepend ............................................................................................ 194
Figure 115: Inbound Route - Multiple Mode .............................................................................................. 195
Figure 116: Blacklist Configuration Parameters ........................................................................................ 196
Figure 117: Blacklist csv File ..................................................................................................................... 197
Figure 118: Conference Invitation From web GUI ..................................................................................... 200
Figure 119: Conference Recording ........................................................................................................... 202
Figure 120: Conference Schedule ............................................................................................................ 206
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Figure 121: Click On Prompt To Create IVR Prompt ................................................................................ 209
Figure 122: Record New IVR Prompt ....................................................................................................... 210
Figure 123: Upload IVR Prompt ................................................................................................................ 210
Figure 124: Language Settings for Voice Prompt ..................................................................................... 212
Figure 125: Voice Prompt Package List .................................................................................................... 212
Figure 126: New Voice Prompt Language Added ..................................................................................... 213
Figure 127: Upload Single Voice Prompt for Entire Language Pack ........................................................ 214
Figure 128: Voicemail Email Settings ....................................................................................................... 218
Figure 129: Voicemail Group..................................................................................................................... 219
Figure 130: Ring Group ............................................................................................................................. 220
Figure 131: Ring Group Configuration ...................................................................................................... 222
Figure 132: Sync LDAP Server option ...................................................................................................... 223
Figure 133: Manually Sync LDAP Server ................................................................................................. 223
Figure 134: Ring Group Remote Extension .............................................................................................. 223
Figure 135: Paging/Intercom Group .......................................................................................................... 224
Figure 136: Page/Intercom Group Settings .............................................................................................. 225
Figure 137: Call Queue ............................................................................................................................. 226
Figure 138: Agent Login Settings .............................................................................................................. 229
Figure 139: Edit Extension Group ............................................................................................................. 230
Figure 140: Select Extension Group in Outbound Route .......................................................................... 231
Figure 141: Edit Pickup Group .................................................................................................................. 232
Figure 142: Edit Pickup Feature Code ...................................................................................................... 233
Figure 143: Music On Hold Default Class ................................................................................................. 234
Figure 144: Configure Analog Trunk without Fax Detection ..................................................................... 238
Figure 145: Configure Extension For Fax Machine .................................................................................. 238
Figure 146: Configure Extension for Fax Machine: Analog Settings ........................................................ 239
Figure 147: Configure Inbound Rule for Fax............................................................................................. 239
Figure 148: Create Fax Extension ............................................................................................................ 240
Figure 149: Inbound Route to Fax Extension ........................................................................................... 240
Figure 150: Create Follow Me ................................................................................................................... 243
Figure 151: Edit Follow Me ....................................................................................................................... 243
Figure 152: Configure One-Key Dial ......................................................................................................... 246
Figure 153: One-Key Dial Destinations ..................................................................................................... 247
Figure 154: Create New DISA................................................................................................................... 248
Figure 155: Create New Event List ........................................................................................................... 253
Figure 156: Create Dial By Name Group .................................................................................................. 255
Figure 157: Dial By Name Group In IVR Key Pressing Events ................................................................ 256
Figure 158: Dial By Name Group In Inbound Route ................................................................................. 257
Figure 159: Configure Extension First Name and Last Name .................................................................. 258
Figure 160: Status->PBX Status->Active Calls - Ringing ......................................................................... 259
Figure 161: Status->PBX Status->Active Calls – Call Established ........................................................... 259
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Figure 162: call connection less than half hour......................................................................................... 260
Figure 163: call connection between half an hour and one hour .............................................................. 260
Figure 164: call connection more than one hour....................................................................................... 261
Figure 165: Configure to Monitor an Active Call ....................................................................................... 261
Figure 166: Enable/Disable Feature codes ............................................................................................... 268
Figure 167: Download Recording File from CDR Page ............................................................................ 268
Figure 168: Create New PIN Group .......................................................................................................... 275
Figure 169: PIN members ......................................................................................................................... 276
Figure 170: Outbound PIN ........................................................................................................................ 276
Figure 171: CDR Record........................................................................................................................... 277
Figure 172: Fax Sending in Web UI .......................................................................................................... 285
Figure 173: Announcements Center ......................................................................................................... 286
Figure 174: Announcements Center Group Configuration........................................................................ 287
Figure 175: Announcements Center Code Configuration ......................................................................... 288
Figure 176: Announcements Center example........................................................................................... 288
Figure 177: Create New Room ................................................................................................................. 290
Figure 178: Room Status .......................................................................................................................... 290
Figure 179: Add batch rooms .................................................................................................................... 291
Figure 180: Create New Wake Up Service ............................................................................................... 291
Figure 181: Wake up Call executed .......................................................................................................... 292
Figure 182: Status->PBX Status ............................................................................................................... 293
Figure 183: Trunk Status ........................................................................................................................... 293
Figure 184: Extension Status .................................................................................................................... 295
Figure 185: Queue Status ......................................................................................................................... 296
Figure 186: Conference Room Status ...................................................................................................... 297
Figure 187: Digital Channels Status ......................................................................................................... 300
Figure 188: Parking Lot Status ................................................................................................................. 301
Figure 189: System Status->Storage Usage ............................................................................................ 303
Figure 190: System Status->Resource Usage ......................................................................................... 303
Figure 191: System Events->Alert Events Lists: Disk Usage ................................................................... 305
Figure 192: System Events->Alert Events Lists: External Disk Usage ..................................................... 305
Figure 193: System Events->Alert Events Lists: Modify Admin Password ............................................... 305
Figure 194: System Events->Alert Events Lists: Memory Usage ............................................................. 306
Figure 195: System Events->Alert Events Lists: System Reboot ............................................................. 306
Figure 196: System Events->Alert Events Lists: System Update ............................................................. 306
Figure 197: System Events->Alert Events Lists: System Crash ............................................................... 307
Figure 198: System Events->Alert Log ..................................................................................................... 307
Figure 199: System Events->Alert Log ..................................................................................................... 308
Figure 200: Filter for Alert Log .................................................................................................................. 308
Figure 201: CDR Filter .............................................................................................................................. 309
Figure 202: Call Report ............................................................................................................................. 310
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Figure 203: Call Report Entry with Audio Recording File .......................................................................... 312
Figure 204: Automatic Download Settings ................................................................................................ 312
Figure 205: CDR Report ........................................................................................................................... 313
Figure 206: Detailed CDR Information ...................................................................................................... 313
Figure 207: Downloaded CDR File Sample .............................................................................................. 313
Figure 208: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ................................ 314
Figure 209: Downloaded CDR File Sample - Source Channel and Dest Channel 2 ................................ 314
Figure 210: CDR Statistics ........................................................................................................................ 315
Figure 211: CDR->Recording Files ........................................................................................................... 316
Figure 212: Network Upgrade ................................................................................................................... 322
Figure 213: Local Upgrade........................................................................................................................ 323
Figure 214: Upgrading Firmware Files ...................................................................................................... 324
Figure 215: Reboot UCM6510 .................................................................................................................. 324
Figure 216: Create New Backup ............................................................................................................... 326
Figure 217: Backup / Restore ................................................................................................................... 326
Figure 218: Local Backup ......................................................................................................................... 327
Figure 219: Data Sync .............................................................................................................................. 328
Figure 220: Restore UCM6510 from Backup File ..................................................................................... 329
Figure 221: Cleaner .................................................................................................................................. 330
Figure 222: Reset and Reboot .................................................................................................................. 331
Figure 223: Ethernet Capture.................................................................................................................... 332
Figure 224: PING ...................................................................................................................................... 333
Figure 225: Traceroute .............................................................................................................................. 333
Figure 226: Troubleshooting Analog Trunks ............................................................................................. 334
Figure 227: E&M Immediate Record Trace............................................................................................... 335
Figure 228: Service Check ........................................................................................................................ 335
Figure 229: Network Status....................................................................................................................... 336
Figure 230: SSH Access ........................................................................................................................... 337
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DOCUMENT PURPOSE
This document describes UCM6510 IP PBX specifications, features and will help you to configure your
system via web UI menu to fully manipulate the supported features. The intended audiences of this
document are device administrators. To learn more about UCM6510 IP PBX features, please visit
http://www.grandstream.com/support to download available how-to guides.
This guide covers following topics:

Product overview

Pickup groups

Installation

Music on hold

Getting started

Fax/T.38

System settings

Asterisk manager interface (AMI)

Provisioning

Follow me

Extensions

One-key dial

Analog trunks

DISA

Digital trunks

Callback feature

Data trunk

BLF and event list

VoIP trunks

Dial by name

SLA station

Active calls and monitor

Call routes

Call features

Conference bridge

Internal options

Conference schedule

IAX settings

IVR

SIP settings

Language settings for voice prompt

Value-added features

Voicemail

Status and reporting

Ring group

CDR (Call Details Record)

Paging and intercom group

Upgrading and maintenance

Call queue

Backup/restore

Extension groups

Troubleshooting
UCM6510 IP PBX User Manual
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CHANGE LOG
This section documents significant changes from previous versions of the UCM6510 user manual. Only
major new features or major document updates are listed here. Minor updates for corrections or editing are
not documented here.
Firmware Version 1.0.11.27

Added ability to sort extension status on web UI [Extensions]

Added one click enable / disable feature code [Feature Codes]

Added Uruguay time zone support [Auto Time Updating]

Added distinctive ring tone support [Configure Call Queue] / [Configure IVR] / [Create New SIP
Extension]

Added special character support for SFTP client account [Data Sync]

Added destination directory support for data sync [Data Sync]

Added ring group music on hold [Configure Ring Group]

Added CDR multi-email / time condition support [CDR]

Added blacklist anonymous call block [Blacklist Configurations]

Added ability to sort selected extension in Eventlist [Event List]

Added Banned User list for web UI login attempts [Login Settings]

Added Email template support [Email Templates]

Added outbound route country restriction [Country Codes]

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[

Internal Options/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 1.0.10.44

Added Zero Config DP750 support. [Model Templates]

Added Configure framing with “esf” or “d4” in T1/J1 [Table 51: Digital Hardware Configuration
Parameters: T1/J1 - PRI_NET/PRI_CPE]
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Firmware Version 1.0.10.39

Added multiple modes support for inbound route. [Inbound Route: Multiple Mode]

Added option “Enable Inbound Multiple Mode”, “Inbound Default Mode” and “Inbound Mode 1” for
switching inbound route mode via feature code. [Feature Codes]

Added prepending prefix for inbound route. [Inbound Route: Prepend Example]

Added multiple registration per extension. [Multiple Registrations Per Extension]

Added SIP Message support. [SMS Message Support]

Added 100rel option for 100rel support. [Table 96: SIP Settings/ToS]

Added video preview support. [Table 86: Internal Options/Payload]

Added User Portal Page Fax sending support. [User Portal]

Added Fax intelligent routing. [Fax Intelligent Route]

Added Re-Invite with two media (audio, image) support for fax sending. [Fax with Two Media]

Added option “Max Concurrent Sending Fax” in Fax settings. [Configure Fax/T.38]

Added option “Fax Queue Length” in Fax settings. [Configure Fax/T.38]

Added Google Service Setting Support. [Google Service Settings Support]

Added Conference Schedule. [CONFERENCE SCHEDULE]

Added Setup Wizard. [Setup Wizard]

Added ability to customize specific prompt. [Customize Specific Prompt]

Added option “ALL” when making backup file. [Backup/Restore]

Added “Enable Destination” and “Default Destination” in Follow Me settings. [FOLLOW ME]

Added “Call Duration Limit” option in web UI->PBX->Internal Options->General. [Internal
Options/General]

Added “Enable Auto Email To User” option in web UI->PBX->Internal Options->General. [Internal
Options/General]

Added options “ICE Support” and “STUN Server” in web UI->PBX->Internal Options->RTP Settings.
[Internal Options/RTP Settings]

Added payload type setting for VP8 in web UI->PBX->Internal Options->Payload. [Internal
Options/Payload]

Added options “External Host” and “Use IP address in SDP” in web UI->PBX->SIP Settings->NAT. [SIP
Settings/NAT]

Improved CDR. [CDR Improvement]

Added Network Status page under web UI->Maintenance->Troubleshooting->Network Status [Network
Status]
Firmware Version 1.0.2.7

Added PRI T310 configuration [Table 48: Digital Hardware Configuration Parameters: E1 –
PRI_NET/PRI_CPE]

Added Announcement Center [Announcements Center]
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Firmware Version 1.0.2.5

Added option to enable/disable SSH access via LCD or web UI [SSH Access]

Added ability to select voicemail storage (Email + WAV is supported) [Table 71: Voicemail Email
Settings]

Added support to allow remote peer extensions in ring group [Remote Extension in Ring Group]

Added ability to strip and prepend digits in inbound routes [Table 64: Inbound Rule Configuration
Parameters]

Added ability to search extensions on Extension page [

Search and Edit Extension]

Added user portal for users to log in with extension number, access user information, extension
configuration and CDR [User Portal]

Added support to send Fax via web UI [Fax Sending]

Added “Enable LDAP” option to skip the extension from UCM default LDAP phonebook [Table 31: SIP
Extension Configuration Parameters]

Added video RE-INVITE support

Added DDNS Support [DDNS Settings]

Added support for Call Barging using feature codes [Enable Spy]

Added ability to search the CDR by called number [Table 109: CDR Filter Criteria]

Added ability to select the file types for automatic backup [Backup/Restore]

Added automatic backup support on SD Card or USB storage [Backup/Restore]

Added support to skip trunk authentication by time condition [

Table 33: SIP Extension Configuration Parameters – Features]

Added option to send P-Asserted-Identity header in SIP Register Trunk [Table 57: SIP Register Trunk
Configuration Parameters]

Added ability to specify trunks in CDR filters [CDR]

Added ability to use Pattern in Caller Number to filter CDR [CDR]

Added support to send UNREGISTER when VoIP trunk is disabled [Table 57: SIP Register Trunk
Configuration Parameters]

Added LDAP client support [LDAP Client Configurations]

Added option to specify the chronological order to voice mails [Table 69: Voicemail Settings]

Added option to configure whether to skip pressing ½ to accept or reject calls from Follow Me [Table
77: Follow Me Settings]

Added option to specify port range in Port Forwarding configuration [Table 11: UCM6510 Network
Settings->Port Forwarding]

Added ability to go back to IVR menu from Dial By Name by pressing the star key [Dial By Name
Configuration]

Added support to upgrade SIP end device via SD card in Zero Config [Table 26: Global Policy
Parameters – Maintenance]

Added ability to filter alert logs [Alert Log]

Added ability to delete alert logs [Alert Log]

Added NAT option for peer trunk [Table 63: Outbound Route Configuration Parameters]
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
Improved Automatic Download CDR result format [CDR]

Fixed Digital Trunk SS7 signaling mode inbound / outbound call problem

Fixed Asterisk is crashed while using external MCB and CEI
Firmware Version 1.0.1.12

Added Active Calls feature to monitor call status and barge in active calls [ACTIVE CALLS AND
MONITOR]

Added support to disable the trunk for VoIP trunk and analog trunk [Table 56: Create New SIP Trunk]
[Table 45: Analog Trunk Configuration Parameters]

Added RBS support on T1

Added Frame Relay support on Data Trunk [DATA TRUNK]

Added ‘Assign CIC to D-channel’ option on SS7 settings page [Table 49: Digital Hardware Configuration
Parameters: E1 - SS7]

Added ‘First CIC’ option in SS7 configuration [Table 49: Digital Hardware Configuration Parameters: E1
- SS7]

Added ‘D-Chan’ selection for PRI and SS7 in editing digital ports [Table 48: Digital Hardware
Configuration Parameters: E1 – PRI_NET/PRI_CPE] [Table 49: Digital Hardware Configuration
Parameters: E1 - SS7]

Added support for Ring simultaneously feature for extensions [Table 31: SIP Extension Configuration
Parameters]

Added support for Music On Hold selection per extension [Table 31: SIP Extension Configuration
Parameters]

Added support to disable this extension per extension [Table 31: SIP Extension Configuration
Parameters]

Added ability to set personal password for making outbound calls per extension [Table 31: SIP
Extension Configuration Parameters]

Added ‘TEL URI’ configuration for SIP extension/VoIP trunk [Table 31: SIP Extension Configuration
Parameters] [Table 56: Create New SIP Trunk]

Added E&M Immediate and E&M Wink signaling for T1 [Table 53: Digital Hardware Configuration
Parameters: T1-E&M Immediate/E&M Wink]

Renamed the ‘network backup’ settings items to ‘data sync’ [Data Sync]

Added “Download Search Result” in CDR [CDR]

Added office time and holiday setting support [Office Time] [Holiday]

Added time condition for call forward [EXTENSIONS]

Added support to monitor FXO trunk using SLA [SLA STATION]

Added One-Key Dial function [ONE-KEY DIAL]

Added Follow Me support [FOLLOW ME]

Supported external number as the key pressing event of an IVR

Improved APIs for Zero Config templates and settings [PROVISIONING]

Supported GXP16XX, Surveillance and GS_wave models in Zero Config [PROVISIONING]

Added advanced settings for devices discovered in Zero Config [Device Configuration]
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
Added ability to delete multiple recording files at one time [Recording Files]

Added call queue destination if no answer/timeout [Table 75: Call Queue Configuration Parameters]

Added call queue Music on Hold customization [Table 75: Call Queue Configuration Parameters]

Added restricted AMI access [ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)]
Warning: Please do not enable AMI on the UCM6510 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 UCM6510 system. Please be cautious when enabling AMI access on the
UCM6510 and restrict the permission granted to the AMI user. By using AMI on UCM6510 you agree
you understand and acknowledge the risks associated with this.

Added ability to choose the type(s) of files to be cleaned in cleaner [Cleaner]

Added DTMF configuration per SIP trunk [Table 56: Create New SIP Trunk]

Added ability to upload and play ring group announcement [Table 73: Ring Group Parameters]

Added ability to upload and play paging call announcement [PAGING AND INTERCOM GROUP]

Added Alert-info configuration for distinctive ringing on inbound route [Table 64: Inbound Rule
Configuration Parameters]

Added ability to prepend digits/trunk name to inbound calls’ caller ID [Table 64: Inbound Rule
Configuration Parameters]

Modified Static Routes Interface display when network method is changed [Static Routes]
Firmware Version 1.0.0.25

This is the initial version.
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WELCOME
Thank you for purchasing Grandstream UCM6510 IP PBX appliance. The UCM6510 is an innovative IP
PBX appliance for E1/T1/J1 networks that brings enterprise-grade unified communications and security
protection to enterprises, small-to-medium businesses (SMBs), retail environments and residential settings
in an easy-to-manage fashion. Powered by an advanced hardware platform and revolutionary software
functionalities, the UCM6510 offers a breakthrough turnkey solution for converged voice, video, data, fax,
security surveillance, and mobility applications out of the box without any extra license fees or recurring
costs.
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 adapter with the UCM6510 as it may cause damage to the products
and void the manufacturer warranty.
This document is subject to change without notice. The latest electronic version of this user manual is
available for download here:
http://www.grandstream.com/support
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for
any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.
UCM6510 IP PBX User Manual
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PRODUCT OVERVIEW
Feature Highlights

1 GHz quad-core Cortex A9 application processor, large memory (1GB DDR3 RAM, 32GB Flash), and
dedicated high performance multi-core DSP array for advanced voice processing

1 Integrated 1 T1/E1/J1 interface, 2 PSTN trunk FXO ports, 2 analog telephone/Fax FXS ports with
lifeline capability in case of power outage, and up to 50 SIP trunk accounts

Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC), hardware based
caller ID/call progress tone and smart automated impedance matching for various countries

Gigabit network port(s) with integrated PoE, USB, SD card; integrated NAT router with advanced QoS
support

Strong defense against malicious attacks (Fail2ban, Whitelist, Blacklist, alerts, etc.)

Data communication via T1/E1/J1 and data-voice combined communication via T1/E1/J1 with SS7 or
PRI

Supports up to 2000 SIP endpoint registrations, up to 200 concurrent calls (up to 100 SRTP encrypted
concurrent calls), and up to 64 conference attendees

Flexible dial plan, call routing, site peering, call recording (manual and automatic per SIP call and SIP
trunk), central control panel for endpoints, integrated NTP server, and integrated LDAP contact directory

Automated detection and provisioning of IP phones, video phones, ATAs, gateways, SIP cameras, and
other endpoints for easy deployment

Strongest-possible security protection using SRTP, TLS, and HTTPS with hardware encryption
accelerator

Redundant power supply, advanced support for Hot Standby Clustering and High Availability to
minimize system down time (pending)

Automatic export of previous day’s data; periodically cleans up user data
Technical Specifications
Table 1: Technical Specifications
Interfaces
Analog Telephone FXS Ports
2 RJ11 ports (both with lifeline capability in case of power outage)
PSTN Line FXO Ports
2 RJ11 ports (both with lifeline capability in case of power outage)
T1/E1/J1 Interface
1 RJ45 port
Network Interfaces
NAT Router
Dual Gigabit ports (switched or routed) with PoE;
A 3rd Gigabit port for Hot-Standby Clustering
Yes (user configurable)
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Peripheral Ports
LED Indicators
USB, SD
Power ½, PoE, USB, SD, T1/E1/J1, FXS ½, FXO ½, LAN, WAN, Cluster
Heartbeat
LCD Display
128x32 dot matrix graphic LCD with DOWN and OK buttons
Reset Switch
Yes, long press for factory reset and short press for reboot
Voice/Video Capabilities
Voice-over-Packet
Capabilities
Voice and Fax Codecs
LEC with NLP Packetized Voice Protocol Unit, 128ms-tail-length carrier
grade Line Echo Cancellation, Dynamic Jitter Buffer, Modem detection and
auto-switch to G.711
G.711 A-law/U-law, G.722, G.723.1 5.3K/6.3K, G.726, G.729A/B, iLBC,
GSM, AAL2-G.726-32, ADPCM; T.38
Video Codecs
H.264, H.263, H.263+
QoS
Layer 3 QoS, Layer 2 QoS
Signaling and Control
DTMF Methods
In Audio, RFC2833, and SIP INFO
Digital Signaling
PRI, SS7, MFC/R2, E&M
Provisioning Protocol and
Plug-and-Play
TFTP/HTTP/HTTPS, auto-discovery & auto-provisioning of Grandstream IP
endpoints
via
ZeroConfig
(DHCP
Option
66/multicast
SIP
SUBSCRIBE/mDNS), Eventlist between local and remote trunks
TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP,
Network Protocols
SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS, LDAP,
HDLC, HDLC-ETH, PPP, Frame Relay
Disconnect Methods
Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current
Disconnect, Busy Tone
Security
Media
Advanced Defense
SRTP, TLS, HTTPS, SSH
Fail2ban, alert events, Whitelist, Blacklist, strong password based access
control
Physical
Universal Power Supply
Physical
Dimensions
Environmental
Input: 100-240VAC, 50-60Hz; Output: DC+12VDC, 1.5A
Unit Weight: 2.165 Kg;
Package weight: 3.012 Kg
440mm (L) x 185mm (W) x 44mm (H)
Operating: 32 – 113oF / 0 – 45oC, Humidity 10-90% (non-condensing)
Storage: 14 – 140oF / -10 – 60oC, Humidity 10-90% (non-condensing)
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Mounting
Rack mount and Desktop
Additional Features
English, Simplified Chinese, Traditional Chinese, Spanish, French,
Portuguese, German, Russian, Italian, Polish, Czech for web GUI;
Multi-language Support
Customizable IVR/voice prompts for English, Chinese, British English,
German, Spanish, Greek, French, Italian, Dutch, Polish, Portuguese,
Russian, Swedish, Turkish, Hebrew and Arabic
Caller ID
Polarity Reversal/ Wink
Call Center
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 – BT, NTT Japan
(pending)
Yes, with enable/disable option upon call establishment and termination
Multiple configurable call queues, automatic call distribution (ACD) based
on agent skills/availability/busy level, in-queue announcement
Customizable Auto Attendant
Up to 5 layers of IVR (Interactive Voice Response)
Maximum Call Capacity
Up to 2000 registered SIP endpoints, up to 200 concurrent calls
Conference Bridges
Up to 8 bridges, up to 64 simultaneous conference attendees
Call Features
Call park, call forward, call transfer, DND, DISA, ring group, pickup group,
blacklist, paging/intercom and etc

FCC: Part 15 (CFR 47) Class B, Part 68

CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3,
EN60950-1, TBR21, RoHS
Compliance

A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS
60950, AS/ACIF S002

ITU-T K.21 (Basic Level); UL 60950 (power adapter)

T1: TIA-968-B Section 5.2.4

E1: TBR12/TBR13, E1: AS/ACIF
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INSTALLATION
Before deploying and configuring the UCM6510 series, the device needs to be properly powered up and
connected to network. This section describes detailed information on installation, connection and warranty
policy of the UCM6510 series.
Equipment Packaging
Table 2: UCM6510 Equipment Packaging
Main Case
Yes ( x 1)
Power Adapter
Yes ( x 2)
Ethernet Cable
Yes ( x 1)
Wall Mount
Yes ( x 2)
Screws
Yes ( x 6)
Quick Installation Guide
Yes ( x 1)
Connect your UCM6510
Connect The UCM6510
Figure 1: UCM6510 Front View
Figure 2: UCM6510 Back View
Follow the steps below to connect the UCM6510 for initial setup:
1. Connect one end of an RJ-45 Ethernet cable (cable type: straight through) into the WAN port of the
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UCM6510; connect the other end into the uplink port of an Ethernet switch/hub.
2. Connect the 12V DC power adapter into the DC 12V power jack 1 on the back of the UCM6510. Insert
the main plug of the power adapter into a surge-protected power outlet. (Connect the second power
adapter into the DC 12V power jack 2 for failover purpose in case the first one is down).
3. Wait for the UCM6510 to boot up. The LCD in the front will show its hardware information when the
bootup process is done.
4. Once the UCM6510 is successfully connected to the network, the LED indicator for the WAN port in
the front will be in solid green and the LCD shows up the IP address.
Depending on how the UCM6510 is used, users can follow the steps below for optional setup:
1. PSTN Line Connection: connect PSTN lines from the wall jack to the UCM6510 LINE ports (FXO ports).
2. Analog Line Connection: connect analog lines (phone and fax) to the PHONE ports (FXS ports).
3. T1/E1/J1 Line Connection: connect one end of the T1/E1/J1 cable provided from the service provider
into the T1/E1/J1 port of the UCM6510; connect the other end into the T1/E1/J1 wall jack. T1/E1/J1
crossover cable should be used and it’s not provided in the UCM6510 package. Please see T1/E1/J1
crossover cable pin-out in the figure below:
Figure 3: UCM6510 T1/E1/J1 Crossover Cable Pin-out
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Safety Compliances
The UCM6510 series IP PBX complies with FCC/CE and various safety standards. The UCM6510 power
adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM6510
package only. The manufacturer’s warranty does not cover damages to the device caused by unsupported
power adapters.
Warranty
If the UCM6510 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
Networks, contact our Technical Support Team for a RMA (Return Materials Authorization) number before
the product is returned. Grandstream Networks reserves the right to remedy warranty policy without prior
notification.
Warning:
Use the power adapter provided with the UCM6510 series IP PBX. Do not use a different power adapter
as this may damage the device. This type of damage is not covered under warranty.
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GETTING STARTED
The UCM6510 provides LCD interface, LED indication and web GUI configuration interface.

The LCD displays hardware, software and network information. Users could also navigate in the LCD
menu for device information and basic network configuration.

The LED indication at the front of the device provides interface connection and activity status.

The web GUI gives users access to all the configurations and options for UCM6510 setup.
This section provides step-by-step instructions on how to use the LCD menu, LED indicators and web GUI
of the UCM6510. Once the basic settings are done, users could start making calls from UCM6510 extension
registered on a SIP phone as described at the end of this section.
Use The LCD Menu

Default LCD Display
By default, when the device is powered up, the LCD will show device model (e.g., UCM6510), hardware
version (e.g., V1.4A) and IP address. Press “Down” button and the system time will be displayed (e.g.,
2014-10-21 14:20).

Menu Access
Press “OK” button to start browsing menu options. Please see menu options in [Table 3: LCD Menu
Options].

Menu Navigation
Press the “Down” arrow key to browser different menu options. Press the “OK” button to select an entry.

Exit
If “Back” option is available in the menu, select it to go back to the previous menu. For “Device Info”
“Network Info” and “Web Info” which do not have “Back” option, simply press the “OK” button to go back
to the previous menu. Additionally, the LCD will display default idle screen after staying in menu option
for 15 seconds.

LCD Backlight
The LCD backlight will be on upon key pressing. The backlight will go off after the LCD stays in idle for
30 seconds.
The following table shows the LCD menu options.
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Table 3: LCD Menu Options
View Events
Device Info
Network Info
Network Menu

Critical Events

Other Events

Hardware: Hardware version number

Software: Software version number

P/N: Part number

WAN MAC: WAN side MAC address

LAN MAC: LAN side MAC address

Uptime: System up time since the last reboot

WAN Mode: DHCP, Static IP, or PPPoE

WAN IP: IP address

WAN Subnet Mask

LAN IP: IP address

LAN Subnet Mask

WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE

Static Routes Reset: Click to reset the static route setting

Reboot

Factory Reset

LCD Test Patterns
Press “OK” to start. Then press “Down” button to test different LCD patterns.
When done, press “OK” button to exit.

Fan Mode
Select “Auto” or “On”.

LED Test Patterns
Select “All On” “All Off” or “Blinking” and check LED status for USB, SD,
Factory Menu
T1/E1/J1, Phone 1/Phone 2, Line 1/Line 2 ports. After the LED test, select
“Back” in the menu and the device will show the LED actual status again.

RTC Test Patterns
Select “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 LCD idle screen by pressing
“DOWN” button, or from web GUI->System Status->General page. After the
test, reboot the device manually and the device will display the correct time.

Hardware Testing
Select “Test SVIP” to perform SVIP test on the device. This is mainly for
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factory testing purpose which verifies the hardware connection inside the
device. The diagnostic result displays on the LCD after the test is done.
Web Info
SSH Switch

Protocol: Web access protocol. HTTP or HTTPS. By default, it’s HTTPS

Port: Web access port number. By default it’s 8089

Enable SSH: Enable SSH access.

Disable SSH: Disable SSH access.
By default the SSH access is disabled.
Use The LED Indicators
The UCM6510 has LED indicators in the front to display connection status. The following table shows the
status definitions.
Table 4: UCM6510 LED INDICATORS
LED Indicator
LED Status
Power 1/Power 2
PoE
LAN
Solid: Connected
WAN
Fast Blinking: Data Transferring
USB
Slow Blinking: Trying to connect
SD
OFF: Not Connected
Phone 1 /Phone 2 (FXS)
Line 1/Line 2 FXO
Solid: Connected and working
Fast Blinking (0.5s on/0.5s off): No cable is
T1/E1/J1
connected; or connected but the link is not
working at all.
Slow Blinking (1s on/1s off): Connected but
the link is only working one-way
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Use The Web GUI
Access Web GUI
The UCM6510 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow users to configure the device through a Web browser such as Microsoft IE (version 8+), Mozilla
Firefox, Google Chrome and etc.
Figure 4: UCM6510 web GUI Login Page
To access the web GUI:
1. Connect the computer to the same network as the UCM6510.
2. Ensure the device is properly powered up and shows its IP address on the LCD.
3. Open a web browser on the computer and enter the IP address in the address bar. The web login page
will display as shown above.
4. Enter the administrator’s login and password to access the web configuration menu. The default
administrator’s username and password is “admin” and “admin”. It is highly recommended to change
the default password after login for the first time.
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Note:
By default, the UCM6510 has “Redirect From Port 80” enabled. Therefore, if users type in the UCM6510 IP
address in the web browser, the web page will be automatically redirected to the page using HTTPS and
port 8089.
For example, if the LCD shows 192.168.40.167, please enter 192.168.40.167 in your web browser and the
web page will be redirected to:
https://192.168.40.167:8089
The option “Redirect From Port 80” can be configured under the UCM6510 web GUI->Settings->HTTP
Server.
Setup Wizard
When the user logs in the UCM6510 web UI for the first time, a setup wizard will guide the user to set up basic
configuration. Configurations in setup wizard includes: Time zone, Change password, Network settings,
Extensions, Trunk and routes.
Figure 5: UCM6510 Setup Wizard
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During the wizard, the user can quit the setup wizard at any time to start over with manual configuration. At
the last step of the wizard, the user will be provided with summary for review, before the configuration is
loaded. Once the setup is completed, the system is ready to go.
Web GUI Configurations
There are four main sections in the web GUI for users to view the PBX status, configure and manage the
PBX.

Status: Displays PBX status, System Status, System Events and CDR.

PBX: To configure extensions, trunks, call routes, zero config for auto provisioning, call features,
internal options, IAX settings, SIP settings, as well as ports configuration for digital trunks.

Settings: To configure network settings, firewall settings, change password, LDAP Server, HTTP
Server, Email Settings, Time Settings and NTP server.

Maintenance: To perform firmware upgrade, backup configurations, cleaner setup, reset/reboot,
syslog setup and troubleshooting.
Web GUI Languages
Currently the UCM6510 web GUI supports the following languages:
English
Simplified Chinese
Traditional Chinese
Spanish
French
Portuguese
Russian
Italian
Polish
German
Users can select the displayed language in web GUI login page, or at the upper right of the web GUI after
logging in.
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Figure 6: UCM6510 web GUI Language
Save And Apply Changes
Click on “Save” button after configuring the web GUI options in one page. After saving all the changes,
make sure click on “Apply Changes” button on the upper right of the web page to submit all the changes. If
the change requires reboot to take effect, a prompted message will pop up for you to reboot the device.
Figure 7: UCM6510 web GUI: Apply Changes
Make Your First Call
Power up the UCM6510 and your SIP end point phone. Connect both devices to the network. Then follow
the steps below to make your first call.
1. Log in the UCM6510 web GUI, go to PBX->Basic/Call Routes->Extensions.
2. Click on “Create New SIP Extension” to create a new extension. You will need User ID, Password and
Voicemail Password information to register and use the extension later.
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3. Register the extension on your phone with the SIP User ID, SIP server and SIP Password information.
The SIP server address is the UCM6510 IP address.
4. When your phone is registered with the extension, dial *97 to access the voicemail box. Enter the
Voicemail Password once you hear “Password” voice prompt.
5. Once successfully logged in to the voicemail, you will be prompted with the Voice Mail Main menu.
6. You are successfully connected to the PBX system now.
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SYSTEM SETTINGS
This section explains configurations for system-wide parameters on the UCM6510. Those parameters
include Network Settings, Firewall, Change Password, LDAP server, HTTP server, Email settings, Time
Settings and NTP Server settings.
User Management
User management is on web GUI->Settings->User Management page. User could create multiple
accounts for different administrators to log in the UCM6510 web GUI. Additionally, the system will
automatically create user accounts along with creating new extensions for extension users to login to the
web UI using their extension number and password. All existing user accounts for web UI login will be
displayed on User Management page as shown in the following figure.
Figure 8: User Management Page Display
User Privileges
Three privilege levels are supported:

Super Admin
-
This is the highest privilege. Super Admin can access all pages on UCM6510 web GUI, change
configuration for all options and execute all the operations.
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-
Super Admin can create, edit and delete one or more users with “Admin” privilege
-
Super Admin can edit and delete one or more users with “Consumer” privilege
-
Super Admin can view operation logs generated by all users.
-
By default, the user account “admin” is configured with “Super Admin” privilege and it’s the only
user with “Super Admin” privilege. The User Name and Privilege level cannot be changed or
deleted.
-
Super Admin could change its own login password on web UI->Settings->Change Password
page.
-
Super Admin could view operations done by all the users in web UI->Settings->User
Management->Operation Log.

Admin
-
Users with “Admin” privilege can only be created by “Super Admin” user.
-
“Admin” privilege users are not allowed to access the following pages:
Maintenance->Upgrade
Maintenance->Backup
Maintenance->Cleaner
Maintenance->Reset/Reboot
Settings->User Management->Operation Log

“Admin” privilege users cannot create new users for login.
Consumer
-
A user account for web UI login is created automatically by the system when a new extension is
created.
-
The user could log in the web UI with the extension number and password to access user
information, extension configuration and CDR of that extension.
Create New Web UI User
When logged in as Super Admin, click on
to create a new account for web UI user. The
following dialog will prompt. Configure the parameters as shown in below table.
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Figure 9: Create New User
Table 5: User Management – Create New User
User Name
User Password
Privilege
Configure a username to identify the user which will be required in web UI login.
Letters, digits and underscore are allowed in the user name.
Configure a password for this user which will be required in web UI login. Letters,
digits and underscore are allowed.
This is the role of the web UI user. Currently only “Admin” is supported when Super
Admin creates a new user.
Department
Fax
Email Address
First Name
Last Name
Enter the necessary information to keep a record for this user.
Home Number
Phone Number
Once created, the Super Admin can edit the users by clicking on
or delete the user by clicking on
.
Figure 10: User Management – New Users
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User Portal
The user could log in web UI user portal using the extension number and password. When there is an
extension created in the UCM6510, the corresponding user account for the extension is automatically
created. The user portal allows limited access including user information, extension configuration and CDR
information of the extension. The login username is the extension number and the password is configured
by Super Admin. The following figure shows the dialog of editing the account information by Super Admin.
The User Name must be the extension number and it’s not configurable.
Figure 11: Edit User Information by Super Admin
The following figure shows an example of login page using extension number 1000 as the username.
Figure 12: User Portal Login
After login, the web UI displays is shown as below.
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Figure 13: User Portal Layout
For the configuration parameter information in each page, please refer [Table 5: User Management – Create
New User] for options in User Portal->Basic Information->User Information page; please refer to
[EXTENSIONS] for options in User Portal->Basic Information->Extension page; please refer to [CDR] for
User Portal->Basic Information->CDR page.
Concurrent Multi-User Login
When there are multiple web UI users created, concurrent multi-user login is supported on the UCM6510.
Multiple users could edit options and have configurations take effect simultaneously. However, if different
users are editing the same option or making the same operation (by clicking on “Apply Changes”), a prompt
will pop up as shown in the following figure.
Figure 14: Multiple User Operation Error Prompt
Operation Log
Super Admin has the authority to view operation logs on UCM6510 web GUI->Settings->User
Management->Operation Log page. Operation logs list operations done by all the web UI users, for
example, web UI login, creating trunk, creating outbound rule and etc. There are 6 columns to record the
operation details “Date”, “User Name”, “IP Address”, “Results”, “Page Operation” and “Specific Operation”.
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Figure 15: Operation Logs
The operation log can be sorted and filtered for easy access. Click on the header of each column to sort.
For example, clicking on “Date” will sort the logs according to operation date and time. Clicking on “Date”
again will reverse the order.
Table 6: Operation Log Column Header
Date
The date and time when the operation is executed.
User Name
The username of the user who performed the operation.
IP Address
The IP address from which the operation is made.
Results
The result of the operation.
Page Operation
Specific Operation
The page where the operation is made. For example, login, logout, delete user,
create trunk and etc.
Click on
to view the options and values configured by this operation.
User could also filter the operation logs by time condition, IP address and/or username. Configure these
conditions and then click on
.
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Figure 16: Operation Logs Filter
The above figure shows an example that operations made by user “support” on device with IP
192.168.40.173 from 2014-11-01 00:00 to 2014-11-06 15:38 are filtered out and displayed.
To delete operation logs, users can perform filtering first and then click on
delete the filtered result of operation logs. Or users can click on
to
to delete all
operation logs at once.
Change Password
After logging in the 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->Settings->User Management-> Change Password page.
2. Enter the old password first.
3. Enter the new password and retype the new password to confirm. The new password has to be at least
4 characters. The maximum length of the password is 16 characters.
4. Configure the Email Address that is used when login credential is lost.
5. Click on “Save” and the user will be automatically logged out.
6. Once the web page comes back to the login page again, enter the username “admin” and the new
password to login.
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Figure 17: Change Password
Enter Old Password
Enter the Old Password for UCM6510
Enter New Password
Enter the New Password for UCM6510
Retype New
Password
Email Address
Retype the New Password for UCM6510
Configure the Email address for UCM6510. In case login credential is lost, Email
address is used to retrieve login credential
Change Binding Email
UCM6510 allows user to configure binding email in case login password is lost. UCM6510 login credential
will be sent to the designated email address. The feature can be found under web UI->Settings->User
Management->Change Binding Email.
Figure 18: Change Binding Email
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Table 7: Change Binding Email option
Enter the password
of the account
Email Address
Enter the current login user credential for UCM6510
Email Address is used to retrieve password when password is lost
Network Settings
After successfully connecting the UCM6510 to the network for the first time, users could log in the web GUI
and go to Settings->Network Settings to configure the network parameters for the device. Select each
tab in web GUI->Settings->Network Settings page to configure LAN/WAN settings, 802.1X and Port
Forwarding.
Note:
To connect the UCM6510 to network, T1/E1/J1 data trunk can also be used, instead of using the WAN/LAN
port. Please see section [DATA TRUNK] to use UCM6510 data trunk to connect the device to Internet.
Basic Settings
Please refer to the following tables for basic network configuration parameters on the UCM6510.
Table 8: UCM6510 Network Settings->Basic Settings
Select “Route”, “Switch” or “Dual” mode on the network interface of UCM6510. The
default setting is “Route”.

Route
WAN port interface will be used for uplink connection. LAN port interface will be
used to serve as router.
Method

Switch
WAN port interface will be used for uplink connection. LAN port interface will be
used as bridge for PC connection.

Dual
Both ports can be used for uplink connection. Users will need assign LAN 1 or
LAN 2 as the default interface in option “Default Interface” and configure
“Gateway IP” for this interface if static IP is used for the interface.
Preferred DNS Server
Enter the preferred DNS server address. If Preferred DNS is configured, the
UCM6510 will use it as Primary DNS server.
WAN (when “Method” is set to “Route”)
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IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
DNS Server 1
Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
User Name
Enter the user name to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS
Assign the VLAN tag of the layer 2 QoS packets for WAN port.
802.1Q/VLAN Tag
The default value is 0.
Layer 2 QoS 802.1p
Assign the priority value of the layer 2 QoS packets for WAN port.
Priority Value
The default value is 0.
LAN (when Method is set to “Route”)
IP Address
Enter the IP address assigned to LAN port. The default setting is 192.168.2.1.
Subnet Mask
Enter the subnet mask. The default setting is 255.255.255.0.
DHCP Server Enable
Enable or disable DHCP server capability. The default setting is “Yes”.
DNS Server 1
Enter DNS server address 1. The default setting is 8.8.8.8.
DNS Server 2
Enter DNS server address 2. The default setting is 208.67.222.222.
Allow IP Address
From
Allow IP Address To
Default IP Lease
Time
Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.
Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.
Enter the IP lease time (in seconds). The default setting is 43200.
LAN (when Method is set to “Switch”)
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
DNS Server 1
Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
User Name
Enter the user name to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS
Assign the VLAN tag of the layer 2 QoS packets for LAN port.
802.1Q/VLAN Tag
The default value is 0.
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Layer 2 QoS 802.1p
Assign the priority value of the layer 2 QoS packets for LAN port. The default value
Priority Value
is 0.
LAN 1 / LAN 2 (when Method is set to “Dual”)
If “Dual” is selected as “Method”, users will need assign the default interface to be
Default Interface
LAN 1 (mapped to UCM6510 WAN port) or LAN 2 (mapped to UCM6510 LAN port)
and then configure network settings for LAN 1 and LAN 2. The default interface is
LAN 2.
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Gateway IP
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 interface. The default setting is 0.0.0.0.
DNS Server 1
Enter the DNS server 1 address for static IP settings. The default setting is 0.0.0.0.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
User Name
Enter the user name to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS
Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is
802.1Q/VLAN Tag
0.
Layer 2 QoS 802.1p
Assign the priority value of the layer 2 QoS packets for LAN port. The default value
Priority Value
is 0.

Method: Route
WAN port interface is used for uplink connection; LAN port interface is used as a router. Please see a
sample diagram below.
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Figure 19: UCM6510 Network Interface Method: Route

Method: Switch
WAN port interface is used for uplink connection; LAN port interface is used as bridge for PC
connection.
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Figure 20: UCM6510 Network Interface Method: Switch

Method: Dual
Both WAN port and LAN port are used for uplink connection. WAN port will be mapped to LAN 1
interface; LAN port will be mapped to LAN 2 interface. Users will need assign LAN 1 or LAN 2 as the
default interface in option “Default Interface” and configure “Gateway IP” if static IP is used for this
interface.
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Figure 21: UCM6510 Network Interface Method: Dual
DHCP Client List
This feature can bind MAC to IP address on the LAN port.
When devices receive IP addresses from UCM6510 LAN port, they will be listed on the webUI under “Settings >
Network Settings > DHCP Client List” as shown below.
Figure 22: DHCP Client List
User can bind manually a MAC to an IP address by clicking on
, the following
figure will pop up.
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Figure 23: Add MAC Address Bind
User needs to set the device MAC address and the IP that will be bound to it (the IP address needs to be
within the UCM6510 DHCP range).
In order to bind a batch of listed MAC addresses, user needs to check first the MAC addresses to bind and
click on
. A confirmation popup will be shown, click
to bind
the addresses.
Figure 24: Batch Add MAC Address Bind
After Clicking “OK” to confirm the binding, the “Bind Status” will change from “Unbind” to “Binding”.
802.1X
IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication
mechanism to device before the device is allowed to access Internet or other LAN resources. The UCM6510
supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure show
UCM6510 uses 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.
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Figure 25: UCM6510 Using 802.1X as Client
Figure 26: UCM6510 Using 802.1X EAP-MD5
The following table shows the configuration parameters for 802.1X on UCM6510. Identity and MD5
password are required for authentication, which should be provided by the network administrator obtained
from the RADIUS server. If “EAP-TLS” or “EAP-PEAPv0/MSCHAPv2” is used as the 802.1X mode, users
will also need upload 802.1X CA Certificate and 802.1X Client Certificate, which should be also generated
from the RADIUS server.
Table 9: UCM6510 Network Settings->802.1X
Select 802.1X mode. The default setting is “Disable”. The supported 802.1X mode
are:
802.1X Mode

EAP-MD5

EAP-TLS

EAP-PEAPv0/MSCHAPv2
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Identity
Enter 802.1X mode identity information.
MD5 Password
Enter 802.1X mode MD5 password information.
802.1X CA 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
A static route is a pre-determined path that the network traffic travels to reach a specific host or network.
On the UCM6510, the static route function allows the device to use manually configured routes, rather than
dynamically assigned routes or default gateway configured in the UCM6510 web GUI->Network Settings>Basic Settings to forward traffic. It can be used to define a route when no other routes is available or
necessary, or used in complementary with existing routing on the UCM6510 as a failover backup, and etc.

Click on
to create a new static route. The configuration parameters are listed
in the table below.

Once added, users can select

Select

Static routes configuration can be reset from LCD menu->Network Menu.
to edit the static route.
to delete the static route.
Table 10: UCM6510 Network Settings->Static Routes
Configure the destination IP address or the destination IP subnet for the UCM6510
to reach using the static route.
Destination
Example:
IP address – 192.168.66.4
IP subnet – 192.168.66.0
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 gateway address so that the UCM6510 can reach the destination via
Gateway
this gateway. Gateway address is optional.
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Example:
192.168.40.5
Specify the network interface “LAN”, “WAN” or “Data trunk 1” (“Data Trunk 1”
Interface
option will show only when the data trunk is enabled) on the UCM6510 to reach the
destination using the static route.
The following diagram shows a sample application of static route usage on UCM6510.
Figure 27: UCM6510 Static Route Sample
The network topology of the above diagram is as below:

Network 192.168.69.0 has IP phones registered to UCM6510 LAN 1 address

Network 192.168.40.0 has IP phones registered to UCM6510 LAN 2 address

Network 192.168.66.0 has IP phones registered to UCM6510 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 UCM6510. Therefore, we need configure a
static route on the UCM6510 so that the phones in isolated networks can make calls between each other.
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Figure 28: UCM6510 Static Route Configuration
Port Forwarding
The UCM6510 network interface supports router functions which provides users the ability to do port
forwarding. If the UCM6510 is set to “Route” under web GUI->Settings->Network Settings->Basic
Settings: Method, port forwarding is available for configuration.
The port forwarding configuration is under web GUI->Settings->Network Settings->Port Forwarding
page. Please see related settings in the table below.
Table 11: UCM6510 Network Settings->Port Forwarding
Specify the WAN port number or a range of WAN ports. Up to 8 ports can be
configured.
WAN Port
Note:
When it is set to a range, WAN port and LAN port must be configured with the same
range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from
WAN port will be forwarded to the LAN port with the same port number, for example,
WAN port 1000 will be port forwarding to LAN port 1000.
LAN IP
Specify the LAN IP address.
Specify the LAN port number or a range of LAN ports.
Note:
LAN Port
When it is set to a range, WAN port and LAN port must be configured with the same
range, such as WAN port: 1000-1005 and LAN port: 1000-1005, and access from
WAN port will be forwarded to the LAN port with the same port number, for example,
WAN port 1000 will be port forwarding to LAN port 1000.
Protocol Type
Select protocol type “UDP Only”, “TCP Only” or “TCP/UDP” for the forwarding in the
selected port. The default setting is “UDP Only”.
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The following figures demonstrate a port forwarding example to provide phone’s web UI access to public
side:

The UCM6510 network mode is set to “Route”

The UCM6510 WAN port is connected to uplink switch, with a public IP address configured, e.g. 1.1.1.1.

The UCM6510 LAN port provides DHCP pool that connects to multiple phone devices in the LAN
network 192.168.2.x. The UCM6510 is used as a router, with gateway address 192.168.2.1

There is a GXP2160 connected under the LAN interface network of the UCM6510. It obtains IP address
192.168.2.100 from UCM6510 DHCP pool

On the UCM6510 web UI->Settings->Network Settings->Port Forwarding, configure a port
forwarding entry as the figure shows below.
WAN Port: This is the port opened up on the WAN side for access purpose.
LAN IP: This is the GXP2160 IP address, under the LAN interface network of the UCM6510.
Protocol Type: We select TCP here for web UI access using HTTP.
Figure 29: UCM6510 Port Forwarding Configuration
This will allow users to access the GXP2160 web UI from public side, by typing in address “1.1.1.1:8088”.
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Figure 30: GXP2160 Web Access Using UCM6510 Port Forwarding
DDNS Settings
DDNS setting allows user to access UCM6510 via domain name instead of IP address.
The UCM6510 supports DDNS service from the following DDNS provider:

dydns.org

noip.com

freedns.afraid.org

zoneedit.com

oray.net
Here is an example of using noip.com for DDNS.
1. Register domain in DDNS service provider. Please note the UCM6510 needs to have public IP
access.
Figure 31: Register Domain Name on noip.com
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2. On web UI->Settings->Network Settings->DDNS Settings, enable DDNS service and configure
username, password and host name.
Figure 32: UCM6510 DDNS Setting
3. Now you can use domain name instead of IP address to connect to the UCM6510 web UI.
Figure 33: Using Domain Name to Connect to UCM6510
Firewall
The UCM6510 provides users firewall configurations to prevent certain malicious attack to the UCM6510
system. Users could configure to allow, restrict or reject specific traffic through the device for security and
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bandwidth purpose. The UCM6510 also provides Fail2ban feature for authentication errors in SIP
REGISTER, INVITE and SUBSCRIBE.
To configure firewall settings in UCM6510, go to web GUI->Settings->Firewall page.
Static Defense
Under web GUI->Settings->Firewall->Static Defense page, users will see the following information:

Current service information with port, process and type.

Typical firewall settings.

Custom firewall settings.
The following table shows a sample current service status running on the UCM6510.
Table 12: UCM6510 Firewall->Static Defense->Current Service
Port
Process
Type
7777
Asterisk
TCP/IPv4
389
Slapd
TCP/IPv4
22
Dropbear
TCP/IPv4
80
Lighthttpd
TCP/IPv4
8089
Lighthttpd
TCP/IPv4
69
Opentftpd
UDP/IPv4
9090
Asterisk
UDP/IPv4
6060
zero_config
UDP/IPv4
5060
Asterisk
UDP/IPv4
4569
Asterisk
UDP/IPv4
5353
zero_config
UDP/IPv4
37435
Syslogd
UDP/IPv4
For typical firewall settings, users could configure the following options on the UCM6510.
Table 13: Typical Firewall Settings
Ping Defense
Enable
If enabled, ICMP response will not be allowed for Ping request. The default
setting is disabled. To enable or disable it, click on the check box for the LAN or
WAN interface.
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Ping-of-Death
Defense Enable
Enable to prevent Ping-of-Death attack to the device. The default setting is
disabled. To enable or disable it, click on the check box for the LAN or WAN
interface.
Under “Custom Firewall Settings”, users could create new rules to accept, reject or drop certain traffic going
through the UCM6510. To create new rule, click on “Create New Rule” button and a new window will pop
up for users to specify rule options.
The following figure shows a firewall rule example that will deny SSH access for the UCM6510 from WAN
side.
Figure 34: Create New Firewall Rule
Table 14: Firewall Rule Settings
Rule Name
Specify the Firewall rule name to identify the firewall rule.
Select the action for the Firewall to perform.
Action

ACCEPT

REJECT

DROP
Select the traffic type.

IN
If selected, users will need specify the network interface “LAN”, “WAN” or
Type
“Both” for the incoming traffic.

OUT
Select the service type.
Service

FTP

SSH

Telnet

TFTP

HTTP
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
LDAP

Custom
If selected, users will need specify Source (IP and port), Destination (IP and
port) and Protocol (TCP, UDP or Both) for the service. Please note if the
source or the destination field is left blank, it will be used as “Anywhere”.
The new rule will be listed at the bottom of the page with sequence number, rule name, action, protocol,
type, source, destination and operation. Users can click on
to edit the rule, or click on
to delete
the rule. Save the change and reboot the device for the configuration to take effect.
Dynamic Defense
Dynamic defense can blacklist hosts dynamically when the UCM6510 is set to “Route” under web GUI>Settings->Network Settings->Basic Settings: Method. If enabled, the traffic via TCP connection
coming into the UCM6510 can be monitored, which helps prevent massive connection attempts or brute
force attacks to the device. The blacklist can be created and updated by the UCM6510 firewall, which will
then be displayed in the web page. Please refer to the following table for dynamic defense options on the
UCM6510.
Table 15: UCM6510 Firewall Dynamic Defense
Dynamic Defense
Enable
Enable dynamic defense. The default setting is disabled.
Configure the dynamic defense periodic time interval (in minutes). If the number
Periodical Time
of TCP connections from a host exceeds the “Connection Threshold” within this
Interval
period, this host will be added into Blacklist. The valid value is between 1 and 59
when dynamic defense is turned on. The default setting is 59.
Configure the blacklist update time interval (in seconds). The default setting is
Blacklist Update
120. This defines how long the IP will be blocked once added into the UCM6510
Interval
blacklist. For example, if it’s set to 300 seconds, the blocked IP address will only
be able to establish TCP connection with the UCM6510 again after 300 seconds.
Connection
Threshold
Configure the connection threshold. Once the number of connections from the
same host reaches the threshold during “Periodical Time Interval”, it will be
added into the blacklist. The default setting is 100.
Allowed IPs and ports range, multiple IP addresses and port range.
Dynamic Defense
Whitelist
For example,
192.168.5.100192.168.5.200 1500:2000
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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 UCM6510 within 1
minute, it will be added into UCM6510 blacklist.

This host 192.168.5.7 will be blocked by the UCM6510 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 UCM6510 within 1 minute, it will not be added into UCM6510 blacklist. It can still
establish TCP connection with the UCM6510.
Figure 35: Configure Dynamic Defense
Fail2ban
Fail2Ban feature on the UCM6510 provides intrusion detection and prevention for authentication errors in
SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within “Max Retry Duration”, the
UCM6510 will take action to forbid the host for certain period as defined in “Banned Duration”. This feature
helps prevent SIP brute force attacks to the PBX system.
Table 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 in order to use Fail2Ban for SIP
authentication on the UCM6510.
Banned Duration
Max Retry Duration
Configure the duration (in seconds) for the detected host to be banned. The default
setting is 300. If set to -1, the host will be always banned.
Within this duration (in seconds), if a host exceeds the max times of retry as defined
in “MaxRetry”, the host will be banned. The default setting is 5.
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MaxRetry
Configure the number of authentication failures during “Max Retry Duration” before
the host is banned. The default setting is 10.
Configure IP address, CIDR mask or DNS host in the whiltelist. Fail2Ban will not
Fail2Ban Whitelist
ban the host with matching address in this list. Up to 5 addresses can be added into
the list.
Local Settings
Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make
Asterisk Service
sure both “Enable Fail2Ban” and “Asterisk Service” are turned on in order to use
Fail2Ban for SIP authentication on the UCM6510.
Protocol
Configure the listening port number for the service. Currently only 5060 (for UDP)
is supported.
Configure the number of authentication failures during “Max Retry Duration” before
MaxRetry
the host is banned. The default setting is 10. Please make sure this option is
properly configured as it will override the “MaxRetry” value under “Global Settings”.
Blacklist
Blacklist
Users will be able to view the IPs that have been blocked by UCM.
LDAP Server
The UCM6510 has an embedded LDAP server for users to manage corporate phonebook in a centralized
manner.

By default, the LDAP server has generated the first phonebook with PBX DN “ou=pbx,dc=pbx,dc=com”
based on the UCM6510 user extensions already.

Users could add new phonebook with a different Phonebook DN for other external contacts. For
example, “ou=people,dc=pbx,dc=com”.

All the phonebooks in the UCM6510 LDAP server have the same Base DN “dc=pbx,dc=com”.

“cn” “ou” and “dc” are parts of LDAP data Interchange Format according to RFC 2849, which is how
the LDAP tree is filtered.
Cn= Common Name
ou= Organization Unit
dc= Domain Component

Here is an example of how the search for “ou=pbx,dc=pbx,dc=com” is performed in LDAP server query.
From the dc=com Domain Component, find the dc=pbx Domain Component first. In the dc=pbx Domain
Component, find the Organizational Unit called pbx (ou=pbx) and then find the object that has a
Common Name of admin.
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If users have the Grandstream phone provisioned by the UCM6510, the LDAP directory has been set up
on the phone and can be used right away for users to access all phonebooks generated in the UCM6510.
Additionally, users could manually configure the LDAP client settings to manipulate the built-in LDAP server
on the UCM6510. If the UCM6510 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 UCM6510 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.
To access LDAP Server settings, go to web GUI->Settings->LDAP Server.
LDAP Server Configurations
The following figure shows the default LDAP server configurations on the UCM6510.
Figure 36: LDAP Server Configurations
The UCM6510 LDAP server supports anonymous access (read-only) by default. Therefore the LDAP client
doesn’t have to configure username and password to access the phonebook directory. The “Root DN” and
“Root Password” here are for LDAP management and configuration where users will need provide for
authentication purpose before modifying the LDAP information.
The default phonebook list in this LDAP server can be viewed and edited by clicking on
for the first
phonebook under LDAP Phonebook.
Figure 37: Default LDAP Phonebook DN
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Figure 38: Default LDAP Phonebook Attributes
LDAP Phonebook
Users could use the default phonebook, edit the default phonebook as well as add new phonebook on the
LDAP server. The first phonebook with default phonebook dn “ou=pbx,dc=pbx,dc=com” displayed on the
LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. The contacts
information will need to be modified via web GUI->PBX->Basic/Call Routes->Extensions first. The default
LDAP phonebook will then be updated automatically.

Add new phonebook
A new sibling phonebook of the default PBX phonebook can be added by clicking on “Add” under “LDAP
Phonebook” section.
Figure 39: Add LDAP Phonebook
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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
or select
to edit the phonebook attributes and contact list (see figure below),
to delete the phonebook.
Figure 40: Edit LDAP Phonebook

Import phonebook from your computer to LDAP server
Click on “Import Phonebook” and a dialog will prompt as shown in the figure below.
Figure 41: Import Phonebook
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The file to be imported must be a CSV file with UTF-8 encoding. Users can open the CSV file with
Notepad and save it with UTF-8 encoding.
Here is how a sample file looks like. Please note “Account Number” and “Phonebook DN” fields are
required. Users could export a phonebook file from the UCM6510 LDAP phonebook section first and
use it as a sample to start with.
Figure 42: Phonebook CSV File Format
The Phonebook DN field is the same “Phonebook Prefix” entry as when the user clicks on “Add” to
create a new phonebook. Therefore, if the user enters “phonebook” in “Phonebook DN” field in the
CSV file, the actual phonebook DN “ou=phonebook,dc=pbx,dc=com” will be automatically created by
the UCM6510 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 UCM6510 LDAP Phonebook, the contacts in the CSV file will be
added into the existing phonebook. If the phonebook DN doesn’t exist on the UCM6510 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
UCM6510.
Figure 43: LDAP Phonebook After Import
As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in
LDAP phonebook section, users cannot import contacts with Phonebook DN field “pbx” if existed in
the CSV file.
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
Export phonebook to your computer from UCM6510 LDAP server
Select the checkbox for the LDAP phonebook and then click on “Export Selected Phonebook” to export
the selected phonebook. The exported phonebook can be used as a record or a sample CSV file for
the users to add more contacts in it and import to the UCM6510 again.
Figure 44: Export Selected LDAP Phonebook
LDAP Client Configurations
The configuration on LDAP client is similar when you use other LDAP servers. Here we provide an example
on how to configure the LDAP client on the SIP end points to use the default PBX phonebook.
Assuming the server base dn is “dc=pbx,dc=com”, configure the LDAP clients as follows (case
insensitive):
Base DN: dc=pbx,dc=com
Login DN: Please leave this field empty
Password: Please leave this field empty
Anonymous: Please enable this option
Filter: (|(CallerIDName=%)(AccountNumber=%))
Port: 389
The following figure gives a sample configurations for UCM6510 acting as a LDAP client.
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Figure 45: LDAP Client Configurations
To configure Grandstream IP phones as the LDAP client, please refer to the following example:
Server Address: The IP address or domain name of the UCM6510
Base DN: dc=pbx,dc=com
User Name: Please leave this field empty
Password: Please leave this field empty
LDAP Name Attribute: CallerIDName Email Department FirstName LastName
LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax
LDAP Number Filter: (AccountNumber=%)
LDAP Name Filter: (CallerIDName=%)
LDAP Display Name: AccountNumber CallerIDName
LDAP Version: If existed, please select LDAP Version 3
Port: 389
The following figure shows the configuration information on a Grandstream GXP2200 to successfully use
the LDAP server as configured in [Figure 36: LDAP Server Configurations].
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Figure 46: GXP2200 LDAP Phonebook Configuration
HTTP Server
The UCM6510 embedded web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow the users to configure the PBX through a web browser such as Microsoft IE, Mozilla Firefox
and Google Chrome. By default, the PBX can be accessed directly by typing IP address in the PC’s web
browser (e.g., 192.168.40.50). It will then be automatically redirected to HTTPS using Port 8089 (e.g.,
https://192.168.40.50:8089). Users could also change the access protocol and port as preferred under web
GUI->Settings->HTTP Server.
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Table 17: HTTP Server Settings
Enable or disable redirect from port 80. On the PBX, the default access
Redirect From Port 80
protocol is HTTPS and the default port number is 8089. When this option is
enabled, the access using HTTP with Port 80 will be redirected to HTTPS
with Port 8089. The default setting is “Enable”.
Select HTTP or HTTPS as the protocol to access the HTTP server. The
Protocol Type
default setting is “HTTPS”. This also defines whether to use HTTP or
HTTPS to download the config file in zero config as the UCM6510 is served
as HTTP/HTTPS server that has the device config files for zero config.
Port
Specify port number to access the HTTP server. The default port number is
8089.
Once the change is saved, the web page will be redirected to the login page using the new URL. Enter
the username and password to login again.
Email Settings
Email Settings
The Email application on the UCM6510 can be used to send out alert event Emails, Fax (Fax-To-Email),
Voicemail (Voicemail-To-Email) and etc. The configuration parameters can be accessed via web GUI>Settings->Email Settings.
Table 18: Email Settings
TLS Enable
Enable or disable TLS during transferring/submitting your Email to other
SMTP server. The default setting is “Yes”.

MTA: Mail Transfer Agent. The Email will be sent from the configured
domain. When MTA is selected, there is no need to set up SMTP server
for it or no user login is required. However, the Emails sent from MTA
Type
might be considered as spam by the target SMTP server.

Client: Submit Emails to the SMTP server. A SMTP server is required
and users need login with correct credentials.
Domain
Server
Username
Password
Specify the domain name to be used in the Email when using type “MTA”.
Specify the SMTP server when using type “Client”. For example, if using
Gmail as the SMTP server, you can configure it as smtp.gmail.com:465.
Username is required when using type “Client”. Normally it’s the Email
address.
Password to log in for the above Username (Email address) is required
when using type “Client”.
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Display Name
Sender
Specify the display name in the FROM header in the Email.
Specify the sender’s Email address.
For example, pbx@example.mycompany.com.
The following figure shows a sample Email settings on the UCM6510, assuming the Email is using
smtp.gmail.com as the SMTP server and the port number is 465.
Figure 47: UCM6510 Email Settings
Once the configuration is finished, click on “Save” first. Then click on “Test” button to make sure the Email
setting is working.
The following figure shows the new dialog prompted to test the Email setting. Fill in a valid Email address
to send a test Email to verify the Email settings on the UCM6510.
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Figure 48: UCM6510 Email Settings: Send Test Email
Email Templates
The Email templates on the UCM6510 can be used for email notification the configuration parameters can
be accessed via Web GUI->Settings->Email Settings->Email Templates.
Figure 49: Email Templates
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To configure the email template, simply click the
button under Options column, and edit the template as
desired.
Figure 50: Conference Schedule Template
Time Settings
Auto Time Updating
The current system time on the UCM6510 is displayed on the upper right of the web page. It can also be
found under web GUI->Status->System Status->General.
To configure the UCM6510 to update time automatically, go to web GUI->Settings->Time Settings-> Auto
Time Updating.
Note:
The configurations under Web GUI->Settings->Time Settings->Time Auto Updating page require reboot
to take effect. Please consider configuring auto time updating related changes when setting up the
UCM6510 for the first time to avoid service interrupt after installation and deployment in production.
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Table 19: Auto Time Updating
Specify the URL or IP address of the NTP server for the UCM6510 to
Remote NTP Server
synchronize
the
date
and
time.
The
default
NTP
server
is
ntp.ipvideotalk.com.
If set to “Yes”, the UCM6510 is allowed to get provisioned for Time Zone
Enable DHCP Option 2
from DHCP Option 2 in the local server automatically. The default setting is
“Yes”.
If set to “Yes”, the UCM6510 is allowed to get provisioned for NTP Server
Enable DHCP Option 42
from DHCP Option 42 in the local server automatically. This will override
the manually configured NTP Server. The default setting is “Yes”.
Select the proper time zone option so the UCM6510 can display correct
time accordingly.
Time Zone
If “Self-Defined Tome Zone” is selected, please specify the time zone
parameters in “Self-Defined Time Zone” field as described in below option.
If “Self-Defined Time Zone” is selected in “Time Zone” option, users will
need define their own time zone following the format below.
The syntax is: std offset dst [offset], start [/time], end [/time]
Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0
MTZ+6MDT+5
This indicates a time zone with 6 hours offset and 1 hour ahead for DST,
which is U.S central time. If it is positive (+), the local time zone is west of
Self-Defined Time Zone
the Prime Meridian (A.K.A: International or Greenwich Meridian); If it is
negative (-), the local time zone is east.
M4.1.0,M11.1.0
The 1st number indicates Month: 1, 2, 3…, 12 (for Jan, Feb…Dec.).
The 2nd number indicates the nth iteration of the weekday: (1st Sunday,
3rd Tuesday…). Normally 1, 2, 3, 4 are used. If 5 is used, it means the last
iteration of the weekday.
The 3rd number indicates weekday: 0, 1, 2…6 (for Sun, Mon, Tues… Sat).
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
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Set Time Manually
To manually set the time on the UCM6510, go to Web GUI->Settings->Time Settings->Set Time
Manually. The format is YYYY-MM-DD HH:MI:SS.
Figure 51: Set Time Manually
Note:
Manually setup time will take effect immediately after saving and applying change in the web UI. If users
would like to reboot the UCM6510 and keep the manually setup time setting, please make sure “Remote
NTP Server”, “Enable DHCP Option 2” and “Enable DHCP Option 42” options under Web GUI->Settings>Time Settings->Time Auto Updating page are unchecked or set to empty. Otherwise, time auto updating
settings in this page will take effect after reboot.
Office Time
On the UCM6510, the system administrator can define “office time”, which can be used to configure time
condition for extension call forwarding schedule and inbound rule schedule. To configure office time, go to
Web GUI->Settings->Time Settings->Office Time. Click on “Create New Office Time” to create an office
time.
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Figure 52: Create New Office Time
Table 20: Create New Office Time
Start Time
Configure the start time for office hour.
End Time
Configure the end time for office hour
Week
Select the work days in one week.
Show Advanced Options
Check this options to show advanced options. Once selected, please
specify “Month” and “Day” below.
Month
Select the months for office time.
Day
Select the work days in one month.
Select “Start Time”, “End Time” and the day for the “Week” for the office time. The system administrator can
also define month and day of the month as advanced options. Once done, click on “Save” and then “Apply
Change” for the office time to take effect. The office time will be listed in the web page as the figure shows
below.
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Figure 53: Settings->Time Settings->Office Time

Click on
to edit the office time.

Click on
to delete the office time.

Click on “Delete Selected Office Times” to delete multiple selected office times at once.
Holiday
On the UCM6510, the system administrator can define “holiday”, which can be used to configure time
condition for extension call forwarding schedule and inbound rule schedule. To configure holiday, go to Web
GUI->Settings->Time Settings->Holiday. Click on “Create New Holiday” to create holiday time.
Figure 54: Create New Holiday
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Table 21: Create New Holiday
Name
Specify the holiday name to identify this holiday.
Holiday Memo
Create a note for the holiday.
Month
Select the month for the holiday.
Day
Select the day for the holiday.
Show Advanced Options
Check this option to show advanced options. If selected, please specify the
days as holiday in one week below.
Select the days as holiday in one week.
Week
Enter holiday “Name” and “Holiday Memo” for the new holiday. Then select “Month” and “Day”. The system
administrator can also define days in one week as advanced options. Once done, click on “Save” and then
“Apply Change” for the holiday to take effect. The holiday will be listed in the web page as the figure shows
below.
Figure 55: 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/How_to_use_office_time_and_holiday_UCM61
00.pdf
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NTP Server
The UCM6510 can be used as a NTP server for the NTP clients to synchronize their time with. To configure
the UCM6510 as the NTP server, set “Enable NTP server” to “Yes” under web GUI->Settings->Time
Settings->NTP Server. On the client side, point the NTP server address to the UCM6510 IP address or
host name to use the UCM6510 as the NTP server.
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Recordings Storage
The UCM6510 supports call recordings automatically or manually and the recording files can be saved in
external storage plugged in the UCM6510 or on the UCM6510 locally. To manage the recording storage,
users can go to UCM6510 web GUI->Settings->Recordings Storage page and select whether to store
the recording files in USB Disk, SD card or locally on the UCM6510.
Figure 56: 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 UCM6510. 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 UCM6510 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 UCM6510.
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 57: 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 58: Recording Storage Category
On the UCM6510, recording files are generated and exist in 3 categories: normal call recording files,
conference recording files, and call queue recording files. Therefore, users have the following options
when select the categories to copy the files to the external device:
-
Recording Files: Copy the normal recording files to the external device.
-
Conference: Copy the conference recording files to the external device.
-
Queue: Copy the call queue recording files to the external device.
-
All: Copy all recording files to the external device.
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Login Settings
After the user logs in the UCM6510 web UI, 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 UCM6510 web GUI->Settings->Login Settings page.
The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user doesn’t make
any operation on web UI within the timeout, the user will be logged out automatically. After that, the web UI
will be redirected to the login page and the user will need to enter username and password to log in.
If set to 0, there is no timeout for the web UI login session and the user will not be automatically logged out.
“User max number of try login” can prevent the UCM6510 from brutal force decryption, if this number is
exceeded user IP address will be banned from accessing the UCM for a period of time based on user
configuration, the default value is 5.
“User prevent login time” specify the period of time in minutes an IP will banned from accessing the UCM
if the User max number of try login is exceeded, the default value is 5.
“Login Banned User List” show the list of IP’s banned from the UCM.
“Login White List” User can add a list of IP’s to avoid the above restriction, thus, they can exceed the User
max number of try login
Figure 59: Login Timeout Settings
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Google Service Settings Support
UCM6510 now supports Google OAuth 2.0 authentication. This feature is used for supporting UCM6510
conference scheduling system. Once OAuth 2.0 is enabled, UCM6510 conference system can access Google
calendar to schedule or update conference.
Google Service Settings can be found under web GUI-> Settings-> Google Service Settings-> Google Service
Settings.
Figure 60: 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 UCM6510
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 61: 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 62: 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 63: Google Service: OAuth2.0 login
6. Now UCM6510 is connected with Google Service
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PROVISIONING
Overview
Grandstream SIP Devices can be configured via Web interface as well as via configuration file through
TFTP/HTTP/HTTPS download. All Grandstream SIP devices support a proprietary binary format
configuration file and XML format configuration file. The UCM6510 provides a Plug and Play mechanism to
auto-provision the Grandstream SIP devices in a zero configuration manner by generating XML config file
and having the phone to download it within LAN area. This allows users to finish the installation with ease
and start using the SIP devices in a managed way.
To provision a phone, three steps are involved, i.e., discovery, configuration and provisioning. This section
explains how Zero Config works on the UCM6510. The settings for this feature can be accessed via Web
UI->PBX->Zero Config.
Configuration Architecture for End Point Device
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 64: Zero Config Configuration Architecture for End Point Device
The configuration in model layer and device layer have all the options 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 UCM6510 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 UCM6510 for auto provisioning. Two methods of
auto provisioning are used.
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Figure 65: UCM6510 Zero Config

SIP SUBSCRIBE
When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The
UCM6510 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 UCM6510 and take the
new configuration.

DHCP OPTION 66
This method should be used only when the UCM6510 is set to “Route” mode under web GUI->Settings>Network Settings->Basic Settings: Method. When the phone restarts (by default DHCP Option 66
is turned on), it will send out a DHCP DISCOVER request. The UCM6510 receives it and returns DHCP
OFFER with the config server path URL in the 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 UCM6510.
To start the auto provisioning process, under Web GUI->PBX->Zero Config->Zero Config Settings, fill in
the auto provision information.
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Figure 66: Auto Provision Settings
Table 22: Auto Provision Settings
Enable Zero Config
Enable or disable the Zero Config feature on the PBX. The default setting
is enabled.
By default, this is disabled. If disabled, when SIP device boots up, the
UCM6510 will only send the configuration path to the device when you
have any manual configuration on the device. This manual configuration
includes:
Any configuration under BASIC and CUSTOM page of the device
-
in Zero Config page
Enable Automatic
Configuration Assignment
If any global or model template (expect for the default template) is
-
selected for the assigned device in Zero Config page.
Note:
When disabled, SIP devices can still be provisioned by manually sending
NOTIFY from the UCM6510 which will include the XML config file URL for
the SIP device to download.
Automatically Assign
Extension
If enabled, when the device is discovered, the PBX will automatically
assign an extension within the range defined in “Zero Config Extension
Segment” to the device. The default setting is disabled.
Click on the link “Zero Config Extension Segment” to specify the extension
Zero Config Extension
Segment
range to be assigned if “Automatically Assign Extension” is enabled. The
default range is 5000-6299. Zero Config Extension Segment range can be
defined in web UI->PBX->Internal Options->General page->Extension
Preference section: “Auto Provision Extensions”.
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If enabled, the extension list will be sent out to the device after receiving
Enable Pick Extension
the device’s request. This feature is for the GXP series phones that support
selecting extension to be provisioned via phone’s LCD. The default setting
is disabled.
Click on the link “Pick Extension Segment” to specify the extension list to
Pick Extension Segment
be sent to the device. The default range is 4000 to 4999. Pick Extension
Segment range can be defined in web UI->PBX->Internal Options>General page->Extension Preference section: “Pick Extensions”.
Pick Extension Period
Specify the number of minutes to allow the phones being provisioned to
(hour):
pick extensions.
Please make sure an extension is manually assigned to the phone or “Automatically Assign Extension” is
enabled during provisioning. After the configuration on the UCM6510 web GUI, click on “Save” and “Apply
Changes”. Once the phone boots up and picks up the config file from the UCM6510, it will take the
configuration right away.
Discovery
Users could manually discover the device by specifying the IP address or scanning the entire LAN network.
Three methods are supported to scan the devices.

PING

ARP

SIP Message (NOTIFY)
Click on “Auto Discover” under web GUI-> PBX-> Zero Config-> Zero Config, fill in the “Scan Method”
and “Scan IP”. The IP address segment will be automatically filled in based on the network mask detected
on the UCM6510. If users need scan the entire network segment, enter 255 (for example, 192.168.5.255)
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 UCM6510 web
GUI->PBX->Zero Config->Auto Provisioning Settings.
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Figure 67: Auto Discover
The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if
assigned), Version, Vendor, Model, Connection Status, Create Config, Options Edit /Delete /Update
/Reboot /Access Device WebGUI) are displayed in the list.
Figure 68: Discovered Devices
Global Configuration
Global Policy
Global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN
with the UCM6510 no matter what the Grandstream device model it is. It is divided into two levels:

Web UI->PBX->Zero Config->Global Policy
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
Web UI->PBX->Zero Config->Global Templates.

Global Templates configuration has higher priority to Global Policy configuration.
Global Policy can be accessed in web GUI->PBX->Zero Config->Global Policy page. On the top of the
configuration table, users can select category in the “Options” dropdown list to quickly navigate to the
category. The categories are:

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.
Figure 69: Global Policy Categories
Select the checkbox on the left of the parameter you would like to configure to active the dropdown list for
this parameter.
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.
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Time Format
NTP Server
Time Zone
Configure the time display in 12-hour or 24-hour format on the SIP end
device’s LCD.
Configure the URL or IP address of the NTP server. The SIP end device
may obtain the date and time from the server.
Configure the time zone used on the SIP end device.
Table 24: Global Policy Parameters – Phone Settings
Default Call Settings
Dial Plan
Configure the default dial plan rule. For syntax and examples, please refer
to user manual of the SIP devices to be provisioned for more details.
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.
Use # as Dial Key
If set to “Yes”, pressing the number key “#” will immediately dial out the
input digits.
If set to “Yes”, the phone will automatically turn on the speaker phone to
Auto Answer by Call-info
answer incoming calls after a short reminding beep, based on the SIP
Call-Info header sent from the server/proxy.
The default setting is enabled.
NAT Traversal
User Random Port
Configure if NAT traversal mechanism is activated.
If set to “Yes”, this parameter will force random generation of both the
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.
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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
the SIP end device.
Source

If “PBX” is selected, the LDAP configuration built-in from UCM6510
web UI->Settings->LDAP Server will be applied.
Address
Configure the IP address or DNS name of the LDAP server.
Port
Configure the LDAP server port. The default value is 389.
This is the location in the directory where the search is requested to begin.
Base DN
User Name
Password
Number Filter
Name Filter
Version
Example:

dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com
Configure the bind “Username” for querying LDAP servers. The field can
be left blank if the LDAP server allows anonymous binds.
Configure the bind “Password” for querying LDAP servers. The field can be
left blank if the LDAP server allows anonymous binds.
Configure the filter used for number lookups. Please refer to user manual
for more details.
Configure the filter used for name lookups. Please refer to user manual for
more details.
Select the protocol version for the phone to send the bind requests. The
default value is 3.
Specify the “name” attributes of each record which are returned in the
LDAP search result.
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
Display Name
can be displayed.
Example:

Max Hits
%cn %sn %telephoneNumber
Specify the maximum number of results to be returned by the LDAP server.
Valid range is 1 to 3000. The default value is 50.
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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. The default value
is 30.
Sort Results
Incoming Calls
Outgoing Calls
Lookup Display Name
Specify whether the searching result is sorted or not. The default setting is
No.
Configure to enable LDAP number searching when receiving calls. The
default setting is No.
Configure to enable LDAP number searching when making calls. The
default setting is No.
Configures the display name when LDAP looks up the name for incoming
call or outgoing call. It must be a subset of the LDAP Name Attributes.
XML Phonebook
Select the source of the phonebook XML server.

Disable
Disable phonebook XML downloading.

Manual
Once selected, users need specify downloading protocol HTTP,
HTTPS or TFTP and the server path to download the phonebook XML
file. The server path could be IP address or URL, with up to 256
Phonebook XML Server
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, the entries added
entries on download
manually will be automatically removed.
Table 26: Global Policy Parameters – Maintenance
Upgrade and Provision
Firmware source via ZeroConfig provisionoing could a URL for external
server address, local UCM directory or USB media if plugged in to the
UCM6510.
Firmware Source
Select a source to get the firmware file:

URL
If select to use URL to upgrade, complete the configuration for the
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following four parameters: “Upgrade Via”, “Server Path”, “File Prefix”
and “File Postfix”.

Local UCM Server
Firmware can be uploaded to the UCM6510 internal storage for
firmware upgrade. If selected, click on “Manage Storage” icon next to
“Directory” option, upload firmware file and select directory for the end
device to retrieve the firmware file.

Local USB Media
If selected, the USB storage device needs to be plugged into the
UCM6510 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 UCM6510 and the
firmware file must be put under a folder named “ZC_firmware” in the
USB storage root directory.
Upgrade via
Server Path
When URL is selected as firmware source, configure upgrade via TFTP,
HTTP or HTTPS.
When URL is selected as firmware source, configure the firmware
upgrading server path.
When URL is selected as firmware source, configure the firmware file
File Prefix
prefix. If configured, only the firmware with the matching encrypted prefix
will be downloaded and flashed into the phone, if URL is selected as
firmware source.
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.
Allow DHCP Option 43/66
If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can
be redirected.
If enabled, the end point device will automatically upgrade if a new firmware
is detected. Users can select automatic upgrading by day, by week or by
minute.

By week
Once selected, specify the day of the week to check HTTP/TFTP
Automatic Upgrade
server for firmware upgrades or configuration files changes.

By day
Once selected, specify the hour of the day to check the HTTP/TFTP
server for firmware upgrades or configuration files changes.

By minute
Once selected, specify the interval X that the SIP end device will
request for new firmware every X minutes.
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Firmware Upgrade Rule
Specify how firmware upgrading and provisioning request to be sent.
Web Access
Admin Password
Configure the administrator password for admin level login.
End-User Password
Configure the end-user password for the end user level login.
Web Access Mode
Select HTTP or HTTPS as the web access protocol.
Web Server Port
Configure the port for web access. The valid range is 1 to 65535.
Security
Enable Telnet/SSH access for the SIP end device. If the SIP end device
Disable Telnet/SSH
supports Telnet access, this option controls the Telnet access of the device;
if the SIP end device supports SSH access, this option controls the SSH
access of the device.
Syslog
Syslog Server
Configure the URL/IP address for the syslog server.
Syslog Level
Select the level of logging for syslog.
Send SIP Log
Configure whether the SIP log will be included in the syslog message.
Table 27: Global Policy Parameters – Network Settings
Basic Settings
Configure how the SIP end device shall obtain the IP address. DHCP or
PPPoE can be selected.

DHCP
Once selected, users can specify the Host Name (option 12) of the SIP
end device as DHCP client, and Vendor Class ID (option 60) used by
IP Address
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
Layer 3 QoS
Define the Layer 3 QoS parameter. This value is used for IP Precedence,
Diff-Serv or MPLS. Valid range is 0-63.
Layer 2 QoS Tag
Assign the VLAN Tag of the Layer 2 QoS packets. Valid range is 0 -4095.
Layer 2 QoS Priority Value
Assign the priority value of the Layer 2 QoS packets. Valid range is 0-7.
STUN Server
Configure the IP address or Domain name of the STUN server. Only nonsymmetric 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.
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Table 28: Global Policy Parameters – Customization
Wallpaper
Check this option if the SIP end device shall use 1024 x 600 resolution for
the LCD screen wallpaper.

Configure the location where wallpapers are stored.
Screen Resolution 1024 x
600
Source

File
If “URL” is selected as source, specify the URL of the wallpaper file. If
“Local UCM Server” is selected as source, click to upload wallpaper
file to the UCM6510.
Check this option if the SIP end device shall use 800 x 400 resolution for
the LCD screen wallpaper.

Screen Resolution 800 x 400
Source
Configure the location where wallpapers are stored.

File
If “URL” is selected as source, specify the URL of the wallpaper file. If
“Local UCM Server” is selected as source, click to upload wallpaper
file to the UCM6510.
Check this option if the SIP end device shall use 480 x 272 resolution for
the LCD screen wallpaper.

Screen Resolution 480 x 272
Source
Configure the location where wallpapers are stored.

File
If “URL” is selected as source, specify the URL of the wallpaper file. If
“Local UCM Server” is selected as source, click to upload wallpaper
file to the UCM6510.
Check this option if the SIP end device supports 320 x 240 resolution for
the LCD screen wallpaper.

Screen Resolution 320 x 240
Source
Configure the location where wallpapers are stored.

File
If “URL” is selected as source, specify the URL of the wallpaper file. If
“Local UCM Server” is selected as source, click to upload wallpaper
file to the UCM6510.
Global Templates
Global Templates can be accessed in web GUI->PBX->Zero Config->Global Templates. Users can create
multiple global templates with different sets of configurations and save the templates. Later on, when the
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user configures the device in Edit Device dialog->Advanced Settings, the user can select to use one of the
global template for the device. Please refer to section [Manage Devices] for more details on using the global
templates.
When creating global template, users can select the categories and the parameters under each category
to be used in the template. The global policy and the selected global template will both take effect when
generating the config file. However, the selected global template has higher priority to the global policy
when it comes to the same setting option/field. If the same option/field has different value configured in the
global policy and the selected global template, the value for this option/field in the selected global template
will override the value in global policy.

Click on “Create New Template” to add a global template. Users will see the following configurations.
Table 29: Create New Template
Template Name
Create a name to identify this global template.
Description
Provide a description for the global template. This is optional.
Active
Check this option to enable the global template.

Click on
to edit the global template.
The window for editing global template is shown in the following figure. In the “Options” field, after
entering the option name key word, the options containing the key word will be listed. Users could then
select the options to be modified and click on “Add Option” to add it into the global template.
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Figure 70: 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
option from the template. On the right side of each option, users can click on
to remove this
to reset the option
value to the default value.

Click on “Save” to save this global template.

The created global templates will show in the web UI->PBX->Zero Config->Global Templates page.
Users can click on
to delete the global template or click on “Delete Selected Templates” to delete
multiple selected templates at once.

Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected
templates.
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Model Configuration
Model Templates
Model layer configuration allows users to apply model-specific configurations to different devices. Users
could create/edit/delete a model template by accessing web GUI, page PBX->Zero Config->Model
Templates. If multiple model templates are created and enabled, when the user configures the device in
Edit Device dialog->Advanced Settings, the user can select to use one of the model template for the device.
Please refer to section [Manage Devices] for more details on using the model template.
For each created model template, users can assign it as default model template. If assigned as default
model template, the values in this model template will be applied to all the devices of this model. There is
always only one default model template that can be assigned at one time on the UCM6510.
The selected model template and the default model template will both take effect when generating the
config file for the device. However, the model template has higher priority to default model template when
it comes to the same setting option/field. If the same option/field has different value configured in the default
model template and the selected model template, the value for this option/field in the selected model
template will override the value in default model template.

Click on “Create New Template” to add a model template.
Table 30: Create New Model Template
Select a model to apply this template. The supported Grandstream models
Model
are listed in the dropdown list for selection.
Template Name
Create a name for the model template.
Description
Enter a description for the model template. This is optional.
Select to assign this model template as the default model template. The
Default Model Template
value of the option in default model template will be overridden if other
selected model template has a different value for the same option.
Check this option to enable the model template.
Active

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.
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Figure 71: Edit Model Template
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/sites/default/files/Resources/config-template_0.zip.

Click on Save when done. The model template will be displayed on web UI->PBX->Zero Config>Model Templates page.

Click on
to delete the model template or click on “Delete Selected Templates” to delete multiple
selected templates at once.

Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected
model templates.
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Model Update
UCM6510 zero config feature supports provisioning all models of Grandstream SIP end devices. Templates
for most of the Grandstream models are built in with the UCM6510 already. Templates for GXP16XX,
GSWave and Surveillance products requires user to download on UCM6510 web UI->PBX->Zero Config>Model Update first as those products are more often used as needed. After downloading the model
template, 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 UCM6510.
Figure 72: Template Management
In case the UCM6510 is placed in the private network and Internet access is restricted, users will not be
able to get packages by downloading and installing from the remote server. Model template package can
be manually uploaded from local device through web UI. Please contact Grandstream customer support if
the model package is needed for manual uploading.
Figure 73: Upload Model Template Manually
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Device Configuration
On web GUI, page PBX->Zero Config->Zero Config, users could create new device, delete existing
device(s), make special configuration for a single device, or send NOTIFY to existing device(s).
Create New Device
Besides configuring the device after the device is discovered, users could also directly create a new device
and configure basic settings before the device is discovered by the UCM6510. Once the device is plugged
in, it can then be discovered and provisioned. This gives the system administrator adequate time to set up
each device beforehand.
Click on “Create New Device” and the following dialog will show. Follow the steps below to create the
configurations for the new device.
1. Firstly, select a model for the device to be created and enter its MAC address, IP address and firmware
version (optional) in the corresponding field.
2. Basic settings will show a list of settings based on the model selected in step 1. Users could assign
extensions to accounts, assign functions to Line Keys and Multiple-Purposed Keys if supported on the
selected model.
3. Click on “Create New Device” to save the configuration for this device.
Figure 74: Create New Device
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Manage Devices
The device manually created or discovered from Auto Discover will be listed in the web UI->PBX->Zero
Config->Zero Config page. Users can see the devices with their MAC address, IP address, vendor, model
and etc.
Figure 75: Manage Devices
1. Click on
2. Click on
to access the web UI of the phone.
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 fivelevel structure.
Figure 76: 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
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in all levels will take effect for the device. If there are same options existing in different level
configurations with different value configured, the higher level configuration will override the lower level
configuration.
(1) Global Policy
This is the lowest level configuration. The global policy configured in web UI->PBX->Zero Config>Global Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page PBX>Zero Config->Global Policy.
(2) Global Templates
Select a global template to be used for the device and click on
to add. Multiple global templates
can be selected and users can arrange the priority by adjusting orders via
and
. All the
selected global templates will take effect. If the same option exists on multiple selected global
templates, the value in the template with higher priority will override the one in the template with
lower priority. Click on
to remove the global template from the selected list.
(3) Default Model Template
Default Model Template will be applied to the devices of this model. Default model template can be
configured in model template under web UI->PBX->Zero Config->Model Templates page. Please
see default model template option in [Table 30: Create New Model Template].
(4) Model Templates
Select a model template to be used for the device and click on
to add. Multiple global templates
can be selected and users can arrange the priority by adjusting orders via
and
. All the
selected model templates will take effect. If the same option exists on multiple selected model
templates, the value in the template with higher priority will override the one in the template with
lower priority. Click on
to remove the model template from the selected list.
(5) Customize Device Settings
This is the highest level configuration in the “Advanced” settings of the device. Click on “Modify
Customize Device Settings” and following dialog will show.
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Figure 77: Edit Customize Device Settings
Scroll down in the dialog to view and edit the device-specific options. If the users would like to
add more options which are not in the pre-defined list, click on “Add New Field” to add a P value
number and the value to the configuration. The following figure shows setting P value “P1362” to
“en”, which means the display language on the LCD is set to English.
The warning information next to the P value field indicates that the option matching the P value
number exists in the configuration already.
For P value information of different models, please refer to configuration template here
http://www.grandstream.com/sites/default/files/Resources/config-template_0.zip.
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Figure 78: Add P Value in Customize Device Settings

Select multiple devices that need to be modified and then click on
to batch
modify devices.
If selected devices are of the same model, the configuration dialog is like the following figure.
Configurations in five levels are all available for users to modify.
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Figure 79: Modify Selected Devices–Same Model
If
on
selected devices are of different models, the configuration dialog is like the following figure. Click
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 80: Modify Selected Devices—Different Models
Warning:
Perform batch operation will override all the configurations made when editing a single device. For
example, if the user configures a GXP2140 to use template “TempA” in Global Templates level by
editing a single device and then selects several devices including that GXP2140 to batch modify
devices selecting “TempB” in Global Templates level, the user will see the global templates changed
to “TempB” when viewing the configurations for the GXP2140.
After the above configurations, save the changes and go back to web UI->PBX->Zero Config->Zero
Config page. Users could then click on
to send NOTIFY to the SIP end point device and trigger the
provisioning process. The device will start downloading the generated configuration file from the URL
contained in the NOTIFY message.
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Figure 81: Device List in Zero Config
In this web page, users can also click on “Reset All Extensions” to reset the extensions of all the devices.
Sample Application
Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1
GXV3275 phone used by customer support supervisor. 3 of the 8 customer support members speak
Spanish and the rest speak English. We could deploy the following configurations to provisioning the office
phones for the customer support team.
1. Go to web GUI->PBX->Zero Config->Auto Provision Settings, select “Enable Zero Config”.
2. Go to web GUI->PBX->Zero Config->Global Policy, configure Date Format, Time Format and
Firmware Source as follows.
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Figure 82: Zero Config Sample – Global Policy
3. Go to web GUI->PBX->Zero Config->Model Templates, create a new model template “English
Support Template” for GXP2140. Add option “Language” and set it to “English”. Then select the option
“Default Model Template” to make it the default model template.
4. Go to web GUI->PBX->Zero Config->Model Templates, create another model template “Spanish
Support Template” for GXP2140. Add option “Language” and set it to “Español”.
5. After 9 devices are powered up and connected to the LAN network, use “Auto Discover” function or
“Create New Device” function to add the devices to the device list on web UI->PBX->Zero Config>Zero Config.
6. On web GUI->PBX->Zero Config->Zero Config page, users could identify the devices by their MAC
addresses or IP addresses displayed on the list. Click on
to edit the device settings.
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
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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 83: Zero Config Sample – Device Preview 1
8. For the 3 phones used by Spanish support, in “Basic” settings select an available extension for account
1 and click on “Save”. Then click on “Advanced” settings tab to bring up the following dialog.
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Figure 84: Zero Config Sample – Device Preview 2
Select “Spanish Support Template” in “Model Template”. The preview of the device settings is displayed
on the right side and we can see the language is set to “Español” since Model Template has the higher
priority for the option “Language”, which overrides the value configured in default model template.
9. For the GXV3275 used by the customer support supervisor, select an available extension for account
1 on “Basic” settings and click on “Save”. Users can see the preview of the device configuration in
“Advanced” settings. There is no model template configured for GXV3275.
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Figure 85: Zero Config Sample – Device Preview 3
10. Click on “Apply Changes” to apply saved changes.
11. On the web UI->PBX->Zero Config->Zero Config page, click on
to send NOTIFY to trigger the
device to download config file from UCM6510.
Now all the 9 phones in the network will be provisioned with an unique extension registered on the
UCM6510. 3 of the phones will be provisioned to display Spanish on LCD and the other 5 will be provisioned
to display English on LCD. The GXV3275 used by the supervisor will be provisioned to use the default
language on LCD display since it’s not specified in the global policy.
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EXTENSIONS
Create New User
Create New SIP Extension
To manually create new SIP user, go to web GUI->PBX->Basic/Call Routes->Extensions. Click on
“Create New User”->”Create New SIP Extension” and a new dialog window will show for users to fill in the
extension information. The configuration parameters are as follows.
Figure 86: Create New Device
SIP extension options are divided into four categories:

Basic Settings

Media

Features

Specific Time
Click on the tag to view or edit options belonging to that category.
The configuration parameters are as follows.
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Table 31: SIP Extension Configuration Parameters – Basic Settings
General
Extension
The extension number associated with the user.
Configure the CallerID Number that would be applied for outbound calls
from this user.
CallerID Number
Note:
The ability to manipulate your outbound Caller ID may be limited by your
VoIP provider.
Assign permission level to the user. The available permissions are
“Internal”, “Local”, “National” and “International” from the lowest level to
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 in order 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.
Support Hot-Desking Mode
Auth ID
Enable Voicemail
If enabled, SIP Password will accept only alphabet characters and digits;
AuthID will be changed to the same as Extension.
Configure the authentication ID for the user. If not configured, the
extension number will be used for authentication.
Enable voicemail for the user. The default setting is “Yes”.
Configure voicemail password (digits only) for the user to access the
Voicemail Password
voicemail box. A random numeric password is automatically generated. It
is recommended to use the random generated password for security
purpose.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
If selected, this extension will be disabled on the UCM6510.
Disable This Extension
Note:
The disabled extension still exists on the PBX but can’t be used on the end
device.
User Settings
First Name
Last Name
Configure the first name of the user. The first name can contain characters,
letters, digits and _.
Configure the last name of the user. The last name can contain characters,
letters, digits and _.
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Fill in the Email address for the user. Voicemail will be sent to this Email
Email Address
address.
Configure the password for user portal access. A random numeric
password is automatically generated. It is recommended to use the
User Password
randomly generated password for security purpose.
Select the voice prompt language to be used for this extension. The default
setting is “Default” which is the selected voice prompt language under web
GUI->PBX->Internal Options->Language. The dropdown list shows all
the current available voice prompt languages on the UCM6510. To add
Language
more languages in the list, please download voice prompt package by
selecting “Check Prompt List” under web UI->PBX->Internal Options>Language.
Concurrent Registrations
The maximum endpoints which can be registered into this extension. For
security concerns, the default value is 1.
Table 32: SIP Extension Configuration Parameters – Media
SIP Settings
Use NAT when the UCM6510 is on a public IP communicating with devices
NAT
hidden behind NAT (e.g., broadband router). If there is one-way audio
issue, usually it’s related to NAT configuration or Firewall’s support of SIP
and RTP ports. The default setting is enabled.
By default, the UCM6510 will route the media steams from SIP endpoints
through itself. If enabled, the PBX will attempt to negotiate with the
Can Direct Media
endpoints to route the media stream directly. It is not always possible for
the UCM6510 to negotiate endpoint-to-endpoint media routing. The
default setting is “No”.
Select DTMF mode for the user to send DTMF. The default setting is
“RFC2833”. If “Info” is selected, SIP INFO message will be used. If
DTMF Mode
“Inband” is selected, 64-kbit PCMU and PCMA are required. When “Auto”
is selected, RFC2833 will be used if offered, otherwise “Inband” will be
used.
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.
Enable Keep-alive
If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is “Yes”.
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Configure the Keep-alive interval (in seconds) to check if the host is up.
Keep-alive Frequency
The default setting is 60 seconds.
Configure the Alert-Info, when UCM6510 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
detected and the received Fax will be sent to the Email address
Fax Mode
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
This option controls how the extension can be used on devices within
different types of network.

Allow All
Device in any network can register this extension.

Strategy
Local Subnet Only
Only the user in specific subnet can register this extension. Up to three
subnet addresses can be specified.

A Specific IP Address
Only the device on the specific IP address can register this extension.
The default setting is “Allow All”.
Select audio and video codec for the extension. The available codecs are:
PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, ILBC,
Codec Preference
ADPCM, H.264, H.263 and H.263p.
Table 33: SIP Extension Configuration Parameters – Features
Call Transfer
Configure the Call Forward Unconditional target number. If not configured,
Call Forward Unconditional
the Call Forward Unconditional feature is deactivated. The default setting
is deactivated.
Select time condition for Call Forward Unconditional. CFU takes effect only
during the selected time condition. The available time conditions are
“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
CFU Time Condition
Office Time or Holiday” and “Specific”.
Note:

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.
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
Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
Configure the Call Forward No Answer target number. If not configured,
Call Forward No Answer
the Call Forward No Answer feature is deactivated. The default setting is
deactivated.
Select time condition for Call Forward No Answer. The available time
conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday” and “Specific”.
Note:

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.
CFN Time Condition

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
Call Forward Busy
Configure the Call Forward Busy target number. If not configured, the Call
Forward Busy feature is deactivated. The default setting is deactivated.
Select time condition for Call Forward Busy. The available time conditions
are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.
CFB Time Condition

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
CC Settings
If enabled, UCM6510 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’s disabled.
Two modes for Call Completion are supported:
CC Mode

Normal: This extension is used as ordinary extension.

For Trunk: This extension is registered from a PBX.
The default setting is “Normal”.
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Configure the maximum number of CCSS agents which may be allocated
CC Max Agents
for this channel. In other words, this 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
CC Max Monitors
created for this device. In other words, this number tells how many callers
may request CC services for a specific device at one time. The minimum
value is 1.
Ring Simultaneously
Enable this option to have an external number ring simultaneously along
Ring Simultaneously
with the extension. If a register trunk is used for outbound, the register
number will be used to be displayed for the external number as caller ID
number.
External Number
Set the external number to be rang simultaneously. ‘-’ is the connection
character which will be ignored.
Time Condition for Ring
Ring the external number simultaneously along with the extension on the
Simultaneously
basis of this time condition.
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
UCM6510, which can be configured in the global ring timeout setting under
Ring Timeout
web GUI->Internal Options->IVR Prompt: General Preference. The valid
range is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
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.
Skip Trunk Auth
·
If set to ‘By Time’, users can skip entering the password when making
outbound calls during the selected time condition.
·
If set to ‘No’, users will be asked to enter the password when making
outbound calls.
Time Condition for Skip
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.
Support Hot-Desking Mode
If enabled, SIP Password will accept only alphabet characters and digits.
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Auth ID will be changed to the same as Extension.
Enable LDAP
If enabled, the extension will be added to LDAP Phonebook PBX list.
Enable WebRTC Support
Enable registration and call from Web RTC.
Specify which Music On Hold class to suggest to the bridged channel when
Music On Hold
putting them on hold.
The maximum duration of call-blocking.
Call Duration Limit
Table 34: SIP Extension Configuration Parameters – Specific Time
Specific Time
Click to add Time Condition to configure specific time for this extension.
Time Condition
Create New IAX Extension
The UCM6510 supports Inter-Asterisk eXchange (IAX) protocol. IAX is used for transporting VoIP telephony
sessions between servers and terminal devices. IAX is similar to SIP but also has its own characteristic.
For more information, please refer to RFC5465.
To manually create new IAX user, go to web GUI->PBX->Basic/Call Routes->Extensions. Click on
“Create New IAX Extension” and a new dialog window will show for users to fill in the extension information.
The configuration parameters are as follows.
Table 35: IAX Extension Configuration Parameters – Basic Settings
General
Extension
The extension number associated with the user.
Configure the CallerID Number that would be applied for outbound calls
from this 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 in order 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.
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Enable voicemail for the user. The default setting is “Yes”.
Enable Voicemail
Configure voicemail password (digits only) for the user to access the
voicemail box. A random numeric password is automatically generated. It
Voicemail Password
is recommended to use the random generated password for security
purpose.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
If selected, this extension will be disabled on the UCM6510.
Disable This Extension
Note:
The disabled extension still exists on the PBX but can’t be used on the end
device.
User Settings
First Name
Last Name
Email Address
Configure the first name of the user. The first name can contain characters,
letters, digits and _.
Configure the last name of the user. The last name can contain characters,
letters, digits and _.
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
Configure the password for user portal access. A random numeric
User Password
password is automatically generated. It is recommended to use the
randomly generated password for security purpose.
Select the voice prompt language to be used for this extension. The default
setting is “Default” which is the selected voice prompt language under web
GUI->PBX->Internal Options->Language. The dropdown list shows all
Language
the current available voice prompt languages on the UCM6510. To add
more languages in the list, please download voice prompt package by
selecting “Check Prompt List” under web UI->PBX->Internal Options>Language.
Table 36: IAX Extension Configuration Parameters – Media
SIP 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
Require Call Token
lock out users who depend on backward compatibility when peer
authentication credentials are shared between physical endpoints. The
default setting is “Yes”.
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Enable SRTP for the call. The default setting is disabled.
SRTP
Select Fax Mode. The default setting is “None”.

None: Disable Fax. This is the default setting.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
Fax Mode
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
This option controls how the extension can be used on devices within
different types of network.

Allow All
Device in any network can register this extension.

Strategy
Local Subnet Only
Only the user in specific subnet can register this extension. Up to three
subnet addresses can be specified.

A Specific IP Address
Only the device on the specific IP address can register this extension.
The default setting is “Allow All”.
Select audio and video codec for the extension. The available codecs are:
PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, ILBC,
Codec Preference
ADPCM, H.264, H.263 and H.263p.
Table 37: IAX Extension Configuration Parameters – Features
Call Transfer
Configure the Call Forward Unconditional target number. If not configured,
Call Forward Unconditional
the Call Forward Unconditional feature is deactivated. The default setting
is deactivated.
Select time condition for Call Forward Unconditional. CFU takes effect only
during the selected time condition. The available time conditions are
“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:
CFU Time Condition

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
Call Forward No Answer
Configure the Call Forward No Answer target number. If not configured,
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the Call Forward No Answer feature is deactivated. The default setting is
deactivated.
Select time condition for Call Forward No Answer. The available time
conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday” and “Specific”.
Note:

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.
CFN Time Condition

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
Call Forward Busy
Configure the Call Forward Busy target number. If not configured, the Call
Forward Busy feature is deactivated. The default setting is deactivated.
Select time condition for Call Forward Busy. The available time conditions
are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.
CFB Time Condition

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
Ring Simultaneously
Enable this option to have an external number ring simultaneously along
Ring Simultaneously
with the extension. If a register trunk is used for outbound, the register
number will be used to be displayed for the external number as caller ID
number.
External Number
Set the external number to be rang simultaneously. ‘-’ is the connection
character which will be ignored.
Time Condition for Ring
Ring the external number simultaneously along with the extension on the
Simultaneously
basis of this time condition.
Other Settings
Configure the number of seconds to ring the user before the call is
Ring Timeout
forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6510, which can be configured in the global ring timeout setting under
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web GUI->Internal Options->IVR Prompt: General Preference. The valid
range is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
Enable automatic recording for the calls using this extension. The default
setting is disabled. The recording files can be accessed under web GUI-
Auto Record
>CDR->Recording Files.

If set to “Yes”, users can skip entering the password when making
outbound calls.

Skip Trunk Auth
If set to “By Time”, users can skip entering the password when making
outbound calls during the selected time condition.

If set to “No”, users will be asked to enter the password when making
outbound calls.
Time Condition for Skip
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.
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
Music On Hold
putting them on hold.
The maximum duration of call-blocking.
Call Duration Limit
Table 38: IAX Extension Configuration Parameters – Specific Time
Specific Time
Time Condition
Click to add Time Condition to configure specific time for this extension.
Create New FXS Extension
The UCM6510 supports Foreign eXchange Subscriber (FXS) interface. FXS is used when user needs to
connect analog phone lines or FAX machines to the UCM6510.
To manually create new FXS user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on
“Create New User”->”Create New FXS Extension” and a new dialog window will show for users to fill in the
extension information. The configuration parameters are as follows.
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Table 39: FXS Extension Configuration Parameters – Basic Settings
General
Extension
The extension number associated with the user.
Analog Station
Select the FXS port to be assigned for this extension.
Configure the CallerID Number that would be applied for outbound calls
from this 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 in order to make outbound calls using this rule.
Enable Voicemail
Enable voicemail for the user. The default setting is “Yes”.
Configure voicemail password (digits only) for the user to access the
Voicemail Password
voicemail box. A random numeric password is automatically generated. It
is recommended to use the random generated password for security
purpose.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
If selected, this extension will be disabled on the UCM6510.
Disable This Extension
Note:
The disabled extension still exists on the PBX but can’t be used on the end
device.
User Settings
First Name
Last Name
Email Address
Configure the first name of the user. The first name can contain characters,
letters, digits and _.
Configure the last name of the user. The last name can contain characters,
letters, digits and _.
Fill in the Email address for the user. Voicemail will be sent to this Email
address.
Configure the password for user portal access. A random numeric
User Password
password is automatically generated. It is recommended to use the
randomly generated password for security purpose.
Language
Select the voice prompt language to be used for this extension. The default
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setting is “Default” which is the selected voice prompt language under web
GUI->PBX->Internal Options->Language. The dropdown list shows all
the current available voice prompt languages on the UCM6510. To add
more languages in the list, please download voice prompt package by
selecting “Check Prompt List” under web UI->PBX->Internal Options>Language.
Table 40: FXS Extension Configuration Parameters – Media
Analog Settings
Call Waiting
User ‘#’ as SEND
RX Gain
TX Gain
Configure to enable/disable call waiting feature. The default setting is “No”.
If configured, the # key can be used as SNED key after dialing the number
on the analog phone. The default setting is “Yes”.
Configure the RX gain for the receiving channel of analog FXS port. The
valid range is -30Db to +6Db. The default setting is 0.
Configure the TX gain for the transmitting channel of analog FXS port. The
valid range is -30Db to +6Db. The default setting is 0.
Configure the minimum period of time (in milliseconds) that the hook-flash
MIN RX Flash
must remain unpressed for the PBX to consider the event as a valid flash
event. The valid range is 30ms to 1000ms. The default setting is 200ms.
Configure the maximum period of time (in milliseconds) that the hook-flash
MAX RX Flash
must remain unpressed for the PBX to consider the event as a valid flash
event. The minimum period of time is 256ms and it can’t be modified. The
default setting is 1250ms.
If enabled, a polarity reversal will be marked as received when an outgoing
call is answered by the remote party. For some countries, a polarity
Enable Polarity Reversal
reversal is used for signaling the disconnection of a phone line and the call
will be considered as hang up on a polarity reversal. The default setting is
“Yes”.
Specify “ON”, “OFF” or a value (the power of 2) from 32 to 1024 as the
number of taps of cancellation.
Echo Cancellation
Note:
When configuring the number of taps, the number 256 is not translated
into 256ms of echo cancellation. Instead, 256 taps mean 256/8 = 32 ms.
The default setting is “ON”, which is 128 taps.
3-Way Calling
Configure to enable/disable 3-way calling feature on the user. The default
setting is enabled.
Send CallerID After
Configure the number of rings before sending CID. Default setting is 1.
Fax Mode
For FXS extension, there are three options available in Fax Mode. The
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default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.

Fax Gateway: If selected, the UCM6510 can support conversation
and processing of Fax data from T.30 to T.38 or T.38 to T.30. This
feature is only available for FXS or FXO port.
Table 41: FXS Extension Configuration Parameters – Features
Call Transfer
Configure the Call Forward Unconditional target number. If not configured,
Call Forward Unconditional
the Call Forward Unconditional feature is deactivated. The default setting
is deactivated.
Select time condition for Call Forward Unconditional. CFU takes effect only
during the selected time condition. The available time conditions are
“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:
CFU Time Condition

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
Configure the Call Forward No Answer target number. If not configured,
Call Forward No Answer
the Call Forward No Answer feature is deactivated. The default setting is
deactivated.
Select time condition for Call Forward No Answer. The available time
conditions are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday” and “Specific”.
Note:
CFN Time Condition

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.
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
Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
Call Forward Busy
Configure the Call Forward Busy target number. If not configured, the Call
Forward Busy feature is deactivated. The default setting is deactivated.
Select time condition for Call Forward Busy. The available time conditions
are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.
CFB Time Condition

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
CC Settings
If enabled, UCM6510 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.
Ring Simultaneously
Enable this option to have an external number ring simultaneously along
Ring Simultaneously
with the extension.
If a register trunk is used for outbound, the register number will be used to
be displayed for the external number as caller ID number.
External Number
Set the external number to be rang simultaneously. ‘-’ is the connection
character which will be ignored.
Time Condition for Ring
Ring the external number simultaneously along with the extension on the
Simultaneously
basis of this time condition.
Hotline
Enable Hotline
Hotline Number
If enabled, hotline dialing plan will be activated, a pre-configured number
will be used according to selected Hotline Type.
Configure the Hotline Number
Configure the Hotline Type:

Immediate Hotline: When the phone is off-hook, UCM6510 will
immediately dial the preset number
Hotline Type

Delay Hotline: When the phone is off-hook, if there is no dialing
within 5 seconds, UCM6510 will dial the preset number.
Other Settings
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Configure the number of seconds to ring the user before the call is
forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6510, which can be configured in the global ring timeout setting under
web GUI->Internal Options->IVR Prompt: General Preference. The valid
Ring Timeout
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 disabled. The recording files can be accessed under web GUI-
Auto Record
>CDR->Recording Files.

If set to “Yes”, users can skip entering the password when making
outbound calls.

Skip Trunk Auth
If set to “By Time”, users can skip entering the password when making
outbound calls during the selected time condition.

If set to “No”, users will be asked to enter the password when making
outbound calls.
Time Condition for Skip
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, this extension will be added to LDAP Phonebook PBX list; if
Enable LDAP
disabled, this extension will be skipped when creating LDAP Phonebook.
Select which Music On Hold class to suggest to extension when putting
Music On Hold
the active call on hold.
Configure the maximum duration of call-blocking.
Call Duration Limit
Table 42: FXS Extension Configuration Parameters – Specific Time
Specific Time
Time Condition
Click to add Time Condition to configure specific time for this extension.
Batch Add Extensions
Batch Add SIP Extensions
In order to add multiple SIP extensions, BATCH add can be used to create standardized SIP extension
accounts. However, unique extension user name can’t be set using BATCH add.
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Under web GUI->PBX->Basic/Call Routes->Extensions, click on “Batch Add Extensions”->”Batch Add
SIP Extensions”.
Table 43: Batch Add SIP Extension Parameters
General
Start Extension
Create Number
Configure the starting extension number of the batch of extensions to be
added.
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 setting is “Internal”.
Permission
Note:
Users need to have the same level as or higher level than an outbound
rule’s privilege in order to make outbound calls using this rule. If the
outbound rule privilege is disabled, this option will not take effect.
Enable Voicemail
Enable Voicemail for the user. The default setting is “Yes”.
Configure the SIP/IAX password for the users. Two options are available to
create password for the batch of extensions.
SIP/IAX Password

User Random Password.
A random secure password will be automatically generated. It is
recommended to use this password for security purpose.

Enter a password to be used on all the extensions in the batch.
Configure Voicemail password (digits only) for the users.

User Random Password.
A random password in digits will be automatically generated. It is
Voicemail Password
recommended to use this password for security purpose.

Enter a password to be used on all the extensions in the batch.
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
UCM6510, which can be configured in the global ring timeout setting under
Ring Timeout
web GUI->Internal Options: 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
Auto Record
setting is disabled. The recording files will be saved in external storage if
plugged in and can be accessed under web GUI->CDR->Recording Files.
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Skip Voicemail Password
Verification
Music On Hold
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
Select which Music On Hold class to suggest to extensions when putting
them on hold.
If enabled, the batch added extensions will be added to LDAP Phonebook
Enable LDAP
PBX list; if disabled, the batch added extensions will be skipped when
creating LDAP Phonebook.
Enable WebRTC Support
Call Duration Limit
If enabled, extensions will be able to login to user portal and use Web RTC
features.
Configure the maximum duration of call-blocking.
SIP Settings
Use NAT when the PBX is on a public IP communicating with devices
NAT
hidden behind NAT (e.g., broadband router). If there is one-way audio issue,
usually it’s related to NAT configuration or Firewall’s support of SIP and RTP
ports. The default setting is enabled.
By default, the PBX will route the media steams from SIP endpoints through
Can Direct Media
itself. If enabled, the PBX will attempt to negotiate with the endpoints to
route the media stream directly. It is not always possible for the 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
DTMF Mode
“RFC2833”. If “Info” is selected, SIP INFO message will be used. If “Inband”
is selected, 64-kbit codec PCMU and PCMA are required. When “Auto” is
selected, RFC2833 will be used if offered, otherwise “Inband” will be used.
Enable Keep-alive
Keep-alive Frequency
If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is “Yes”.
Configure the number of seconds for the host to be up for Keep-alive. The
default 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
TEL URI
attached to the Request-Line and TO header in the SIP request to indicate
the E.164 number. If set to “Enable”, “Tel:” will be used instead of “SIP:” in
the SIP request. The default setting is disabled.
Concurrent Registrations
The maximum endpoints which can be registered into this extension. For
security concerns, the default value is 1.
Other Settings
SRTP
Fax Mode
Enable SRTP for the call. The default setting is “No”.
Select Fax mode for this user. The default setting is “None”.

None: Disable Fax.
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
Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in Fax
setting page under web UI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
This option controls how the extension can be used on devices within
different types of network.

Allow All
Device in any network can register this extension.
Strategy

Local Subnet Only
Only the user in specific subnet can register this extension. Up to three
subnet addresses can be specified.

A Specific IP Address
Only the device on the specific IP address can register this extension.
The default setting is “Allow All”.
Enable T.38 UDPTL
Skip Trunk Auth
Enable or disable T.38 UDPTL Support.
If enabled, users will not need enter the “PIN Set” required by the outbound
rule to make outbound calls. The default setting is “No”.
Select audio and video codec for the extension. The available codecs are:
Codec Preference
PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, ILBC,
ADPCM, LPC10, H.264, H.263 and H.263p. In the selected codec list, users
can click on UP or DOWN arrow to adjust the order for the codec priority.
Batch Add IAX Extensions
Under Web GUI->PBX->Basic/Call Routes->Extensions, click on “Batch Add Extensions”->”Batch Add
IAX Extensions”.
Table 44: Batch Add IAX Extension Parameters
General
Start Extension
Create Number
Configure the starting extension number of the batch of extensions to be
added.
Specify the number of extensions to be added. The default setting is 5.
Assign permission level to the user. The available permissions are
Permission
“Internal”, “Local”, “National” and “International” from the lowest level to
the highest level. The default setting is “Internal”.
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Note:
Users need to have the same level as or higher level than an outbound
rule’s privilege in order to make outbound calls from this rule.
Enable Voicemail
Enable Voicemail for the user. The default setting is “Yes”.
Configure the SIP/IAX password for the users. Three options are available
to create password for the batch of extensions.

User Random Password.
A random secure password will be automatically generated. It is
SIP/IAX Password
recommended to use this password for security purpose.

Use Extension as Password.

Enter a password to be used on all the extensions in the batch.
Configure Voicemail password (digits only) for the users.

User Random Password.
A random password in digits will be automatically generated. It is
Voicemail Password
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 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
UCM6510, which can be configured in the global ring timeout setting under
Ring Timeout
web GUI->Internal Options->IVR Prompt: General Preference. The valid
range is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
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.
Skip Voicemail Password
Verification
Music On Hold
Call Duration Limit
When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
Select which Music On Hold class to suggest to extensions when putting
them on hold.
Configure the maximum duration of call-blocking.
If enabled, the batch added extensions will be added to LDAP Phonebook
Enable LDAP
PBX list; if disabled, the batch added extensions will be skipped when
creating LDAP Phonebook.
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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
Require Call Token
lock out users who depend on backward compatibility when peer
authentication credentials are shared between physical endpoints. The
default setting is “Yes”.
Other Settings
SRTP
Enable SRTP for the call. The default setting is “No”.
Select Fax Mode for this user. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
Fax Mode
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
This option controls how the extension can be used on devices within
different types of network.

Allow All
Device in any network can register this extension.
Strategy

Local Subnet Only
Only the user in specific subnet can register this extension. Up to three
subnet addresses can be specified.

A Specific IP Address.
Only the device on the specific IP address can register this extension.
The default setting is “Allow All”.
If enable “All”, users do not need to enter password when making an
Skip Trunk Auth
outbound call. 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:
Codec Preference
PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, ILBC,
ADPCM, LPC10, H.264, H.263 and H.263p.
Search and Edit Extension
All the UCM6510 extensions are listed under web GUI->PBX->Basic/Call Routes->Extensions, with
status, Extension, CallerID Name, Technology (SIP, IAX and FXS), IP and Port. Each extension has a
checkbox for users to “Modify Selected Extensions” or “Delete Selected Extensions”. Also, options “Edit”
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, “Reboot”
and “Delete”
are available per extension. User can search an extension by
specifying the extension number to find an extension quickly.
Figure 87: Manage Extensions

Status
Users can see the following icon for each extension to indicate the SIP status.

Green:
Free
Blue:
Ringing
Yellow:
In Use
Grey:
Unavailable
Edit single extension
Click on

to start editing the extension parameters.
Reboot the user
Click on
to send NOTIFY reboot event to the device which has an UCM6510 extension already
registered. To successfully reboot the user, “Zero Config” needs to be enabled on the UCM6510 web
GUI->PBX->Basic/Call Routes->Zero Config->Auto Provisioning Settings.

Delete single extension
Click on
to delete the extension. Or select the checkbox of the extension and then click on “Delete
Selected Extensions”.

Modify selected extensions
Select the checkbox for the extension(s). Then click on “Modify Selected Extensions” to edit the
extensions in a batch.

Delete selected extensions
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Select the checkbox for the extension(s). Then click on “Delete Selected Extensions” to delete the
extension(s).
Export Extensions
The extensions configured on the UCM6510 can be exported to csv format file with selected technology
“SIP”, “IAX” or “FXS”. Click on “Import Extensions” scroll down to select “Export Extensions” button and
select technology in the prompt.
Figure 88: Export Extensions
The exported csv file can also serve as a template for users to fill in desired extension information to be
imported to the UCM6510.
Import Extensions
The capability to import extensions to the UCM6510 provides users flexibility to batch add extensions with
similar or different configurations quickly.
1. Export extension csv file from the UCM6510 by clicking on “Export Extensions” button.
2. Fill up the extension information you would like in the exported csv template.
3. Click on “Import Extensions” button. The following dialog will be prompted.
Figure 89: Export Extensions
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4. Select the option in “On Duplicate Extension” to define how the duplicate extension(s) in the imported
csv file should be treated by the PBX.

Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension
information as previously configured without change.

Delete and Recreate: The current extension previously configured will be deleted and the duplicate
extension in the csv file will be loaded to the PBX.

Update Information: The current extension previously configured in the PBX will be kept. However,
if the duplicate extension in the csv file has different configuration for any options, it will override
the configuration for those options in the extension.
5. Click on
to select csv file from local directory in the PC for uploading.
6. Click on “Save” to import the csv file.
7. Click on “Apply Changes” to apply the imported file on the UCM6510.
Email to User
Once the extensions are created with Email address, the PBX administrator can click on button “Email To
User” to send the account registration and configuration information to the user. Please make sure Email
setting under web UI->Settings->Email Settings is properly configured and tested on the UCM6510 before
using “Email To User”.
When click on “Email To User” button, the following message will be prompted in the web page. Click on
OK to confirm sending the account information to all users’ Email addresses.
Figure 90: Email To User: Prompt Information
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The user will receive Email including account registration information and LDAP configuration.
A QR code
is also generated for Mobile applications to scan it and get automatically provisioned. QR code provisioning
is supported on Grandstream Softphone GS Wave AndroidTM application.
Figure 91: Email To User: Account Registration Information and QR Code
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Figure 92: Email To User: LDAP Client Information and QR Code
Multiple Registrations Per Extension
UCM6510 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 UI->PBX>Basic/Call Routes->Edit Extension. The default value is set to 1 for security purpose.
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Figure 94: Extension - Concurrent Registration
SMS Message Support
The UCM6510 provides built-in SIP SMS message support. For SIP end devices such as Grandstream GXP or
GXV phones that supports SIP message, after an UCM6510 account is registered on the end device, the user
can send and receive SMS message. Please refer to the end device documentation on how to send and receive
SMS message.
SMS Message support is a new feature added since firmware 1.0.10.x.
Figure 95: SMS Message Support
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ANALOG TRUNKS
To set up analog trunk on the UCM6510:

Go to web GUI->PBX->Basic/Call Routes->Analog Trunks to add and edit analog trunks.

Go to web GUI->PBX->Ports Config->Analog Hardware to configure analog hardware settings.
Analog Trunks Configuration
Go to web GUI->PBX->Basic/Call Routes->Analog Trunks to add and edit analog trunks.

Click on “Create New Analog Trunk” to add a new analog trunk.

Click on
to edit the analog trunk.

Click on
to delete the analog trunk.
The analog trunk options are listed in the table below.
Table 45: Analog Trunk Configuration Parameters
Channels
Trunk Name
Select the channel for the analog trunk.
Specify a unique label to identify the trunk when listed in outbound routes,
inbound routes and etc.
Enable this option to satisfy two primary use cases, which include emulating
SLA Mode
a simple key system and creating shared extensions on a PBX. Enable SLA
Mode will disable polarity reversal.
The barge option specifies whether or not other stations are allowed to join
Barge Allowed
a call in progress on this trunk. If enabled, the other stations can press the
line button to join the call. The default setting is Yes.
The hold option specifies hold permissions for this trunk. If set to “Open”,
Hold Access
any station can place this trunk on hold and any other station is allowed to
retrieve the call. If set to “Private”, only the station that places the call on
hold can retrieve the call. The default setting is Yes.
Advanced Options
If enabled, a polarity reversal will be marked as received when an outgoing
Enable Polarity Reversal
call is answered by the remote party. For some countries, a polarity reversal
is used for signaling the disconnection of a phone line and the call will be
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considered as “hang up” on a polarity reversal. The default setting is “No”.
When FXO port answers the call, FXS may send a Polarity Reversal. If this
Polarity on Answer Delay
interval is shorter than the value of “Polarity on Answer Delay”, the Polarity
Reversal will be ignored. Otherwise, the FXO will on-hook to disconnect the
call. The default setting is 600ms.
Current Disconnect
Threshold (ms)
This is the periodic time (in ms) that the UCM6510 will use to check on a
voltage drop in the line. The default setting is 200. The valid range is 50 to
3000.
Configure the ring timeout (in ms). Trunk (FXO) devices must have a
timeout to determine if there was a hang up before the line is answered.
Ring Timeout
This value can be used to configure how long it takes before the UCM6510
considers a non-ringing line with hang up activity. The default setting is
8000.
RX Gain
TX Gain
Use CallerID
Configure the RX gain for the receiving channel of analog FXO port. The
valid range is from -13.5 (Db) to + 12.0 (Db). The default setting is 0.
Configure the TX gain for the transmitting channel of analog FXO port. The
valid range is from -13.5 (Db) to + 12.0 (Db). The default setting is 0.
Configure to enable CallerID detection. The default setting is “Yes”.
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
Fax Detection
GUI->PBX->Internal Options->Fax/T.38. The default setting is “No”.
Note:
If enabled, Fax Pass-through cannot be used.
Select the Caller ID scheme for this trunk. If you are not sure which scheme
Caller ID Scheme
to choose, please select “Auto Detect”. The default setting is
“Bellcore/Telcordia”.
FXO Dial Delay(ms)
Configure the time interval between off-hook and first dialed digit for
outbound calls.
Enable automatic recording for the calls using this trunk. The default setting
Auto Record
is disabled. The recording files are saved in external storage device if
plugged in and can be accessed under web GUI->CDR->Recording Files.
Disable This Trunk
If selected, the trunk will be disabled.
This is to implement analog trunk outbound line selection strategy. Three
options are available:
DAHDI Out Line Selection

Ascend
When the call goes out from this analog trunk, it will always try to use
the first idle FXO port. The port order that the call will use to go out
would be port 1->port 2->port 10->port 16. Every time it will start with
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port 1 (if it’s idle).

Poll
When the call goes out from this analog trunk, it will use the port that is
not used last time. And it will always use the port in the order of port 1>2->10->16->1->2->10->16->1->2->10->16…, following the last port
being used.

Descend
When the call goes out from this analog trunk, it will always try to use
the last idle FXO port. The port order that the call will use to go out
would be port 16->port 10->port 2->port 1. Every time it will start with
port 16 (if it’s idle).
The default setting is “Ascend” mode.
Tone Settings
Busy Detection
Busy Detection is used to detect far end hang up or for detecting busy
signal. The default setting is “Yes”.
If “Busy Detection” is enabled, users can specify the number of busy tones
to be played before hanging up. The default setting is 2. Better results might
Busy Tone Count
be achieved if set to 4, 6 or even 8. Please note that the higher the number
is, the more time is needed to hang up the channel. However, this might
lower the probability to get random hang up.
Congestion Detection
Congestion Count
Congestion detection is used to detect far end congestion signal. The
default setting is “Yes”.
If “Congestion Detection” is enabled, users can specify the number of
congestion tones to wait for. The default setting is 2.
Select the country for tone settings. If “Custom” is selected, users could
Tone Country
manually configure the values for Busy Tone and Congestion Tone. The
default setting is “United States of America (USA)”.
Syntax:
f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];
Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000]
Busy 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
Congestion Tone
Syntax:
f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];
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Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000]
Busy Level Range: (-300, 0)
Cadence Range: [0, 16383].
Select Tone Country “Custom” to manually configure Busy Tone value.
Default value:
f1=480@-50,f2=620@-50,c=250/250
Click on “Detect” to detect the busy tone, Polarity Reversal and Current
PSTN Detection
Disconnect by PSTN. Before the detecting, please make sure there are
more than one channel configured and working properly. If the detection
has busy tone, the “Tone Country” option will be set as “Custom”.
PSTN Detection
The UCM6510 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
UCM6510 settings.
1. Go to UCM6510 web GUI->PBX->Basic/Call Routes->Analog Trunks page.
2. Click to edit the analog trunk created for the FXO port.
3. In the dialog window to edit the analog trunk, go to “Tone Settings” section and click on
“PSTN Detection”.
for
Figure 96: UCM6510 FXO Tone Settings
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4. Click on “Detect” to start PSTN detection.
Figure 97: UCM6510 PSTN Detection

If there are two FXO ports connected to PSTN lines, use the following settings for auto-detection.
Detect Model: Auto Detect.
Source Channel: The source channel to be detected.
Destination Channel: The channel to help detecting. For example, the second FXO port.
Destination Number: The number to be dialed for detecting. This number must be the actual PSTN
number for the FXO port used as the destination channel.
Figure 98: UCM6510 PSTN Detection: Auto Detect

If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.
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Figure 99: UCM6510 PSTN Detection: Semi-Auto Detect
Detect Model: Semi-auto Detect.
Source Channel: The source channel to be detected.
Destination Number: The number to be dialed for detecting. This number could be a cell phone
number or other PSTN number that can be reached from the source channel PSTN number.
5. Click “Detect” to start detecting. The source channel will initiate a call to the destination number. For
“Auto Detect”, the call will be automatically answered. For “Semi-auto Detect”, the UCM6510 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 UCM6510
settings.
Table 46: 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
UCM6510 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
UCM6510 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 UCM6510
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”.
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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.
Analog Hardware Configuration
The analog hardware (FXS port and FXO port) on the UCM6510 can be configured under web GUI->PBX>Ports Config->Analog Hardware. 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 100: FXS Ports Signaling Preference
For FXO port, users could manually enter the ACIM settings by selecting the value from dropdown list for
each port. Or users could click on “Detect” for the UCM6510 to automatically detect the ACIM value. The
detecting value will be automatically filled into the settings.
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Figure 101: FXO Ports ACIM Settings
Note:
ACIM setting is very important for the FXO/PSTN line to work properly on the UCM6510. If the users
experience echo, caller ID or disconnecting issue, please make sure to run the ACIM detection to find out
the correct value for impedance setting.
Table 47: Analog Hardware Configuration Parameters
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)”.
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
FXS TISS Override
(TISS). The default setting is No.
If enabled, users can select the impedance value for Two-Wire Impedance
Synthesis (TISS) override. The default setting is 600Ω.
Select the codec to be used for analog lines. North American users should
choose PCMU. All other countries, unless already known, should be
PCMA Override
assumed to be PCMA. The default setting is PCMU.
Note:
This option requires system reboot to take effect.
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Configure whether normal ringing voltage (40V) or maximum ringing
Boost Ringer
voltage (89V) for analog phones attached to the FXS port is required. The
default setting is “Normal”.
Fast Ringer
Low Power
Configure to increase the ringing speed to 25HZ. This option can be used
with “Low Power” option. The default setting is “Normal”.
Configure the peak voltage up to 50V during “Fast Ringer” operation. This
option is used with “Fast Ringer”. The default setting is “Normal”.
If set to “Full Wave”, false ring detection will be prevented for lines where
Ring Detect
Caller ID is sent before the first ring and proceeded by a polarity reversal,
as in UK. The default setting is “Standard”.
Configure the type of Message Waiting Indicator on FXS lines. The default
FXS MWI Mode
setting is “FSK”.

FSK: Frequency Shift Key Indicator

NEON: Light Neon Bulb Indicator.
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DIGITAL TRUNKS
The UCM6510 supports E1/T1/J1 which are physical connection technology used in digital network. T1 is
the North American standard, J1 is used in Japan, whereas E1 is the European standard.
UCM6510 supports four signaling protocols: PRI, MFC/R2, SS7, E&M Immediate and E&M Wink. PRI
provides a varying number of channels depending on the standards in the country of implementation (E1,
T1 or J1); MFC/R2 is a signaling protocol heavily used over E1 trunks; SS7 uses out-of-band signaling,
which travels on a separate, dedicated channel rather than within the same channel as the telephone call,
providing more efficiency and higher security level when the telephone calls are set up. E&M Immediate
and E&M Wink are only valid when using T1 port.
To set up digital trunk on the UCM6510:
1. Go to web UI->PBX->Ports Config->Digital Hardware to configure port type and channels.
2. Go to web UI->PBX->Basic/Call Routes->Digital Trunks to add and edit digit trunk.
3. Go to web UI->PBX->Basic/Call Routes->Outbound Routes and Inbound Routes to configure
outbound and inbound rule for the digital trunk.
Digital Hardware Configuration
Go to web GUI->PBX->Ports Config->Digital Hardware page and configure the following:
3
1
2
Figure 102: Digital Hardware Configuration

Step 1: Click on
to edit digital ports. Please see configuration parameters in the tables below.

Step 2: Click on
to edit group. This assigns channels to be used for the digital port. For E1, 30 B
channels can be assigned to the default group; for T1/J1, 23 B channels can be assigned to the default
group.

Step 3: If fewer than 30 B channels for E1 or 23 B channels for T1/J1 are assigned in default group,
users can click on
to add more groups. This is not necessary in most cases and only default group
is needed.
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Note:
Currently, the group configuration in digit trunks settings is to manage outbound routes only. It doesn’t
control inbound routes. Therefore, if the users have configured multiple groups for the digital trunk, please
make sure the inbound routes for those groups have the same inbound rule configured. Otherwise, inbound
call using the digital trunk might not work properly.
The UCM6510 currently supports E1, T1 and J1 digital hardware type. When different signaling is selected
for E1, T1 or J1, the settings in basic options and advanced options will be different. The following tables
list all the settings to configure digital ports when selecting each signaling.
Table 48: Digital Hardware Configuration Parameters: E1 – PRI_NET/PRI_CPE
Basic Settings
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is used
as the master, the PBX system clock will synchronize to it.
Clock

Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.

Slave: The equipment at the far end of the E1/T1/J1 link is the preferred
source of the master clock.
LBO
RX Gain
TX Gain
Codec
The line build-out (LBO) is the distance between the operators and the PBX.
Please use the default value 0Db unless the distance is long.
Configure the RX gain for the receiving channel of digital port. The valid
range is from -24Db to +12Db.
Configure the TX Gain for the transmitting channel of digital port. The valid
range is -24Db to +12Db.
Select alaw or ulaw. If set to default, alaw will be used for E1.
This configured whether to play the ringback tone from local UCM6510 or
Play Local RBT
not. If enabled, the local UCM6510 will play ringback tone to the caller.
Otherwise, the caller will listen to the tone from peer device. The default
setting is disabled.
Advanced Settings
Switch Type
Select switch type.

EuroISDN: EuroISDN (common in Europe)
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
NI2: National ISDN type 2 (common in the US)

DMS100: Nortel DMS100

4ESS: AT&T 4ESS

5ESS: Lucent 5ESS

NI1: old national ISDN type 1

Q.SIG
Coding
Select “HDB3” or “AMI”.
CRC
Select whether to use CRC4 or not.
This setting is used to specify the type of the callee number. The service
provider will usually verify this. The default setting is “unknown”. In some
very unusual circumstances, you may need set to “Dynamic” or
PRI Dial Plan
“Redundant”.
Note:
When one type is selected, you might not be able to dial another class of
numbers. For example, if “National” is configured, you won’t be able to dial
local or international numbers.
PRI Local Dial Plan
D-Chan
This setting is used to specify the type of the caller number. The service
provider will usually verify this.
Indicates the D channel for control.
International Prefix
National Prefix
Local Prefix
Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each type.
Private Prefix
Unknown Prefix
PRI T310
Configure PRI T310 Timer (in seconds). The default value is 10 seconds.
Select the PRI Indication.

outofband: Use RELEASE, DISCONNECT or other messages with
CAUSE to indicate call progress (e.g., cause: unassigned number or
PRI Indication
user busy).

inband: use in-band tones to play busy or congestion signal to the other
side. This is the default setting.
Reset Interval
The interval that restarts idle channels.
This setting is used to set up the ChannelID in SETUP message. If enabled,
only the specified B channel can be used. Otherwise, select one of the
PRI Exclusive
channels in B channel. If you need override the existing channels selection
routine and force all PRI channels to be marked as exclusively selected,
please enable it.
Facility Enable
If selected, transmission of facility-based ISDN supplementary services
(such as caller name from CPE over facility) will be enabled.
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Configure this option to send overlap digits. If enabled, SETUP message
Overlap Dial
can include some digits of callee number, and rest of the digits can be sent
using INFORMATION message. If disabled, callee number will be sent via
SETUP message when all the digits are ready.
Some switches (AT&T especially) require network specific facility. Currently
NSF
the supported values are "none", "sdn", "megacom", "tollfreemegacom",
"accunet".
Table 49: Digital Hardware Configuration Parameters: E1 - SS7
Basic Settings
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is used
as the master, the PBX system clock will synchronize to it.
Clock

Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.

Slave: The equipment at the far end of the E1/T1 link is the preferred
source of the master clock.
SS7 Variant
Select ITU, ANSI or CHINA.
Originating point code is used to identify the node originating the message,
Originating Point Code
always provided by the operator/ISP.

ITU Format: decimal number.

ANSI & CHINA Format: decimal number or XXX-XXX-XXX.
Destination point code is the address to send the message to, always be
Destination Point Code
provided by the operator/ISP.

ITU Format: decimal number.

ANSI & CHINA Format: decimal number or XXX-XXX-XXX.
When Span Type is E1, ITU & CHINA Range: [0, 4065], ANSI Range: [0,
First CIC
16353].
When Span Type is T1/J1, ITU & CHINA Range: [0, 4072], ANSI Range: [0,
16360].
Assign CIC To D-channel
If set to yes, D-channel will be assigned a CIC. Else, D-channel will not be
assigned. By default, it is set to No.
Network Indicator (NI) should match in nodes, otherwise it might cause
Network Indicator
issues. Users can select "National", "National Spare", "International", or
"International Spare". Usually "National" or "International" is used.
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The line build-out (LBO) is the distance between the operators and the PBX.
LBO
Please use the default value 0dB unless the distance is long.
Configure the RX gain for the receiving channel of digital port. The valid
RX Gain
range is from -24dB to +12dB.
Configure the TX Gain for the transmitting channel of digital port. The valid
TX Gain
range is -24dB to +12dB.
Select alaw or ulaw. If set to default, alaw will be used for E1.
Codec
Advanced Settings
Coding
Select "HDB3" or "AMI".
CRC
Select whether to use CRC4 or not.
Indicates the type of the called number. The receiving switch may use this
Called Nature of Address
indicator during translations to apply the number’s proper dial plan. Users
Indicator
can select
"Unknown",
"Subscriber",
"National", "International" or
"Dynamic".
Indicates the type of the calling number. The receiving switch may use this
Calling Nature of Address
indicator during translations to apply the number’s proper dial plan. Users
Indicator
can select
"Unknown",
"Subscriber",
"National",
"International" or
"Dynamic".
D-chan
Indicates the D channel for control
International Prefix
National Prefix
Configure the prefix in Called Nature of Address Indicator and Calling
Subscriber Prefix
Nature of Address Indicator for each type.
Unknown Prefix
Table 50: Digital Hardware Configuration Parameters: E1 - MFC/R2
Basic Settings
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is used
as the master, the PBX system clock will synchronize to it.
Clock

Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.

Slave: The equipment at the far end of the E1/T1 link is the preferred
source of the master clock.
Variant
MFC/R2 multinational adaption. UCM6510 supports MFC/R2 standards by
ITU and MFC/R2 standards in different countries or regions including
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Argentina, Brazil, China, Czech Republic, Colombia, Ecuador, Indonesia,
Mexico, the Philippines and Venezuela.
If enabled, the callee side will request the caller to send caller number first
and then called number.
Get ANI First
Note:
Options "Get ANI First" and "Skip Category" cannot be enabled at the same
time.
Select the category of the caller. UCM6510 supports four categories:
Category
National Subscriber, National Priority Subscriber, International Subscriber
and International Priority Subscriber.
LBO
RX Gain
TX Gain
The line build-out (LBO) is the distance between the operators and the PBX.
Please use the default value 0dB unless the distance is long.
Configure the RX gain for the receiving channel of digital port. The valid
range is from -24dB to +12dB.
Configure the TX Gain for the transmitting channel of digital port. The valid
range is -24dB to +12dB.
This configured whether to play the ringback tone from local UCM6510 or
Play Local RBT
not. If enabled, the local UCM6510 will play ringback tone to the caller.
Otherwise, the caller will listen to the tone from peer device. The default
setting is disabled.
Advanced Settings
Coding
Select "HDB3" or "AMI".
CRC
Select whether to use CRC4 or not.
MF Back Timeout (ms)
MFC/R2 value in milliseconds for MF timeout. Values smaller than 500ms
are not recommended. -1 represents default value.
MFC/R2 value in milliseconds for the metering pulse timeout. Metering
pulse is sent by some telcos for some R2 variants during a call presumably
Metering Pulse Timeout (ms)
for billing purposes to indicate costs. Should not last more than 500ms, -1
represents default value, and for Argentina the default value is 400ms, for
others is 0ms.
Brazil has a special calling party category for collect calls (llamadas por
cobrar) instead of using the operator (as in Mexico). The R2 spec in Brazil
Alllow Collect Calls
says a special GB tone should be used to reject collect calls.
By default, this is disabled, which means collect calls will be blocked.
Some PBXs require a double-answer process to block collect calls. If users
Double Answer
have problem blocking collect calls using Group B signals, please try
enabling this option.
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Accept On Offer
By default, it’s enabled. In most of cases, this option should be enabled.
If enabled, the callee side will request the caller to send caller category
Skip Category
before sending caller number.
Note:
"Get ANI First" and "Skip Category" cannot be enabled at the same time.
Whether or not report to the other end "accept call with charge". This setting
Charge Calls
has no effect with most telecos. The default setting is enabled
(recommended).
Click on “Custom Options” button (on the right top of the configuration
Custom Options
dialog) and then user can customize desired tone and timer options
accordingly.
Table 51: Digital Hardware Configuration Parameters: T1/J1 - PRI_NET/PRI_CPE
Basic Settings
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is used
as the master, the PBX system clock will synchronize to it.
Clock

Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.

Slave: The equipment at the far end of the E1/T1/J1 link is the preferred
source of the master clock.
LBO
RX Gain
TX Gain
Codec
The line build-out (LBO) is the distance between the operators and the PBX.
Please use the default value 0dB unless the distance is long.
Configure the RX gain for the receiving channel of digital port. The valid
range is from -24dB to +12dB.
Configure the TX Gain for the transmitting channel of digital port. The valid
range is -24dB to +12dB.
Select alaw or ulaw. If set to default, ulaw will be used for T1/J1.
This configured whether to play the ringback tone from local UCM6510 or
Play Local RBT
not. If enabled, the local UCM6510 will play ringback tone to the caller.
Otherwise, the caller will listen to the tone from peer device. The default
setting is disabled.
Framing
Select “esf” or “d4”. Default setting is esf.
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Advanced Settings
Select switch type.
Switch Type
Coding

EuroISDN: EuroISDN (common in Europe)

NI2: National ISDN type 2 (common in the US)

DMS100: Nortel DMS100

4ESS: AT&T 4ESS

5ESS: Lucent 5ESS

NI1: old national ISDN type 1

Q.SIG
Select "B8ZS" or "AMI".
This setting is used to specify the type of the callee number. The service
provider will usually verify this. The default setting is "unknown". In some
very unusual circumstances, you may need set to "Dynamic" or
PRI Dial Plan
"Redundant".
Note:
When one type is selected, you might not be able to dial another class of
numbers. For example, if "National" is configured, you won't be able to dial
local or international numbers.
PRI Local Dial Plan
D-chan
This setting is used to specify the type of the caller number. The service
provider will usually verify this.
Indicates the D channel for control.
International Prefix
National Prefix
Local Prefix
Configure the prefix in PRI Local Dial Plan and PRI Dial Plan for each type.
Private Prefix
Unknown Prefix
PRI T310
Configure PRI T310 Timer (in seconds). The default value is 10 seconds.
Select the PRI Indication.

outofband: Use RELEASE, DISCONNECT or other messages with
CAUSE to indicate call progress (e.g., cause: unassigned number or
PRI Indication
user busy).

inband: use in-band tones to play busy or congestion signal to the other
side. This is the default setting.
Reset Interval
The interval that restarts idle channels.
This setting is used to set up the ChannelID in SETUP message. If enabled,
only the specified B channel can be used. Otherwise, select one of the
PRI Exclusive
channels in B channel. If you need override the existing channels selection
routine and force all PRI channels to be marked as exclusively selected,
please enable it.
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If selected, transmission of facility-based ISDN supplementary services
Facility Enable
(such as caller name from CPE over facility) will be enabled.
Configure this option to send overlap digits. If enabled, SETUP message
can include some digits of callee number, and rest of the digits can be sent
Overlap Dial
using INFORMATION message. If disabled, callee number will be sent via
SETUP message when all the digits are ready.
Some switches (AT&T especially) require network specific facility. Currently
the supported values are "none", "sdn", "megacom", "tollfreemegacom",
NSF
"accunet".
Table 52: Digital Hardware Configuration Parameters: T1/J1 - SS7
Basic Settings
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is used
as the master, the PBX system clock will synchronize to it.
Clock

Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.

Slave: The equipment at the far end of the E1/T1 link is the preferred
source of the master clock.
SS7 Variant
Select ITU, ANSI or CHINA.
Originating point code is used to identify the node originating the message,
Originating Point Code
always provided by the operator/ISP.

ITU Format: decimal number.

ANSI & CHINA Format: decimal number or XXX-XXX-XXX.
Destination point code is the address to send the message to, always be
Destination Point Code
provided by the operator/ISP.

ITU Format: decimal number.

ANSI & CHINA Format: decimal number or XXX-XXX-XXX.
When Span Type is E1, ITU & CHINA Range: [0, 4065], ANSI Range: [0,
First CIC
16353].
When Span Type is T1/J1, ITU & CHINA Range: [0,4072], ANSI Range: [0,
16360].
Assign CIC to D-Channel
Network Indicator
If set to yes, D-channel will be assigned with a CIC. Else, D-channel will not
be assigned with a CIC. By default, it is set to No.
Network Indicator (NI) should match in nodes, otherwise it might cause
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issues. Users can select "National", "National Spare", "International", or
"International Spare". Usually "National" or "International" is used.
The line build-out (LBO) is the distance between the operators and the PBX.
LBO
Please use the default value 0dB unless the distance is long.
Configure the RX gain for the receiving channel of digital port. The valid
RX Gain
range is from -24dB to +12dB.
Configure the TX Gain for the transmitting channel of digital port. The valid
TX Gain
range is -24dB to +12dB.
Codec
Select alaw or ulaw. If set to default, ulaw will be used for T1/J1.
Framing
Select “esf” or “d4”. Default setting is esf.
Advanced Settings
Select "B8ZS" or "AMI".
Coding
Indicates the type of the called number. The receiving switch may use this
Called Nature of Address
indicator during translations to apply the number’s proper dial plan. Users
Indicator
can select
"Unknown",
"Subscriber",
"National", "International" or
"Dynamic".
Indicates the type of the calling number. The receiving switch may use this
Calling Nature of Address
indicator during translations to apply the number’s proper dial plan. Users
Indicator
can select
"Unknown",
"Subscriber",
"National", "International" or
"Dynamic".
Indicates the D channel for control.
D-chan
International Prefix
National Prefix
Configure the prefix in Called Nature of Address Indicator and Calling
Subscriber Prefix
Nature of Address Indicator for each type.
Unknown Prefix
Table 53: Digital Hardware Configuration Parameters: T1-E&M Immediate/E&M Wink
Basic Setting
All E1/T1/J1 spans generate a clock signal on their transmit side. The
parameter determines whether the clock signal from the far end of the
E1/T1/J1 is used as the master source of clock timing. If the far end is used
as the master, the PBX system clock will synchronize to it.
Clock

Master: The port will never be used as a source of timing. This is
appropriate when you know the far end should always be a slave to
you.

Slave: The equipment at the far end of the E1/T1/J1 link is the preferred
source of the master clock.
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Configure the RX gain for the receiving channel of digital port. The valid
RX Gain
range is from -24dB to +12dB.
Configure the TX Gain for the transmitting channel of digital port. The valid
TX Gain
range is -24dB to +12dB.
Codec
Select alaw or ulaw. The default codec is ulaw for T1.
Framing
Select “esf” or “d4”. Default setting is esf.
Advanced Settings
Select B8ZS or AMI. The default setting is B8ZS for T1.
Coding
OutgoingDialDelay
rxwink
The option is only valid for E&M Wink signaling. The dial delay interval after
received WINK event in an outgoing call. The default value is 200ms.
Configure receive wink timing. The default setting is 300ms.
Digital Trunk Configuration
After configuring digital hardware, go to web GUI->PBX->Basic/Call Routes->Digital Trunks.

Click on "Create New Digital Trunk" to add a new digital trunk.

Click on

Click on

Click on
to configure detailed parameters for the digital trunk.
to configure Direct Outward Dialing (DOD) for the digital Trunk.
to delete the digital trunk.
The digital trunk parameters are listed in the table below.
Table 54: Digital Trunk Configuration Parameters
Trunk Name
Configure trunk name to identify the digital trunk.
Channel Group
Configure the digital channel group used by the trunk.
Hide CallerID
Configure to hide outgoing caller ID. The default setting is "No".
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension's CID when the
extension has CID configured. The default setting is "No".
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
Caller ID
the CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine
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which CallerID will be used if they exist:

The CallerID configured for the extension will be looked up first.

If “Keep Trunk CID” is enabled, the CallerID configured for the trunk will
be used.

If the above two are missing, the "Global Outbound CID" defined in web
GUI->PBX->Internal Options->General will be used.
CallerID Name
Configure the new name of the caller when the extension has no CallerID
Name configured.
Enable automatic recording for the calls using this trunk (for SIP trunk only).
Auto Record
The default setting is disabled. The recording files are saved in external
storage device if plugged in and can be accessed under web GUI->CDR>Recording Files.
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
Fax Detection
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
Direct Outward Dialing (DOD) via Digital Trunks
Please refer to section [Direct Outward Dialing (DOD) via VoIP Trunks].
Digital Trunk Troubleshooting
After configuring the digital trunk on the UCM6510 as described above, if it doesn't work as expected,
users can go to capture signaling trace on the UCM6510 web UI for troubleshooting purpose.
Depending on the signaling selected for the digital trunk, users can go to the following pages to capture
trace:
PRI Signaling Trace: web GUI->Maintenance->Troubleshooting ->PRI Signaling Trace
SS7 Signaling Trace: web GUI->Maintenance->Troubleshooting ->SS7 Signaling Trace
MFC/R2 Signaling Trace: web GUI->Maintenance->Troubleshooting ->MFC/R2 Signaling Trace
E&M Trace: web GUI->Maintenance->Troubleshooting->E&M Immediate Record Trace
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Here is the step to capture trace:
1. Click on "Start" to start capturing trace. The output result shows "Capturing..."
2. Once the test is done, click on "Stop" to stop the trace.
3. Click on "Download" to download the trace.
Figure 103: Troubleshooting Digital Trunks
For E&M Immediate Signaling, user could configure “Record Direction” and “Record File Mode”.
After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream
Technical support in the following link for further assistance if the issue is not resolved.
http://www.grandstream.com/support
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DATA TRUNK
The UCM6510 E1/T1/J1 interface also supports data trunk function that allows users to access Internet.
Users can select HDLC, HDLC-ETH, Cisco and PPP protocol for the data trunk.
To use data trunk,
1. Go to web UI->PBX->Ports Config->Digital Hardware page and click
to create a new group.
Designate a channel for data trunk usage in the group setting.
2. Go to web UI->PBX->Basic/Call Routes->Data Trunks page, click on
to edit the data trunk.
3. Save the configuration and click on “Apply Changes” for the change to take effect.
4. Once connected, the data trunk will periodically ping and check the status, with status indicator shown
for the data trunk on the web page. The status indicator shows
if connected successfully.
5. If the user happens to lost connection or experience unstable connection, click on
to reconnect
to help resolve the problem.
6. Users can always click on ON/OFF switch
in the web page to enable/disable the
configured data trunk.
Figure 104: Data Trunk Web Page
Figure 105: Data Trunk Configuration
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Table 55: Data Trunk Configuration Parameters
Data Enable
Select the checkbox to enable/disable data trunk. Users can also click on
the ON/OFF switch in data trunk web page to enable/disable this.
Select the digital channel group from the dropdown list to be used for data
Channel Group
trunk. Users will need create a new group under web UI->PBX->Ports
Config->Digital Hardware page for this purpose.
Encapsulation
Local IP
Subnet Mask
Remote IP
Select the protocol used for the data trunk. The UCM6510 supports HDLC,
HDLC-ETH, PPP, Cisco and Frame Relay.
Configure the local IP address for the data port. This IP address shouldn’t
conflict with the WAN or LAN side IP of the UCM6510.
Configure the subnet mask for the data port.
Configure the remote IP address for the data port. This IP will be the
gateway IP address if “Default Interface” is enabled for the datat trunk.
DNS Server 1
Configure DNS server 1.
DNS Server 2
Configure DNS server 2.
If enabled, this data port will be used as the default interface for Internet
Default Interface
connection. The “Remote IP” will be the gateway IP address. This has
higher priority than the “Default Interface” assignment (LAN 1 or LAN 2)
under web UI->Settings if “Dual” is selected as the network method.
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VOIP TRUNKS
VoIP Trunk Configuration
VoIP trunks can be configured in UCM6510 under web GUI->PBX->Basic/Call Routes->VoIP Trunks.
Once created, the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and Options
to edit/detect the trunk.

Click on "Create New SIP Trunk" or "Create New IAX Trunk" to add a new VoIP trunk.

Click on

Click on
to configure Direct Outward Dialing (DOD) for the SIP Trunk.

Click on
to start LDAP Sync.

Click on
to configure detailed parameters for the VoIP trunk.
to delete the VoIP trunk.
For VoIP trunk example, please refer to the following document:
http://www.grandstream.com/sites/default/files/Resources/how_to_interconnect_multiple_ucm6510_using
_peer_trunk.pdf
The VoIP trunk options are listed in the table below.
Table 56: Create New SIP Trunk
Select the VoIP trunk type.
Type
Provider Name
Host Name

Peer SIP Trunk

Register SIP Trunk
Configure a unique label to identify this trunk when listed in outbound rules,
inbound rules and etc.
Configure the IP address or URL for the VoIP provider’s server of the trunk.
Keep the CID from the inbound call when dialing out. This setting will
Keep Original CID
override “Keep Trunk CID” option. Please make sure that the peer PBX at
the other side supports to match user entry using “username” field from
authentication line.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension’s CID when
the extension has CID configured. The default setting is “No”.
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Turn on this setting when the PBX is using public IP and communicating
NAT
with devices behind NAT. If there is one-way audio issue, usually it is
related to NAT configuration or SIP/RTP port support on the firewall.
If checked, the trunk will be disabled.
Disable This Trunk
Note:
If a current SIP trunk is disabled, UCM6510 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
TEL URI
the Request-Line and TO header in the SIP request to indicate the E.164
number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP
request. The default setting is disabled.
Configure the Caller ID. This is the number that the trunk will try to use
when making outbound calls. For some providers, it might not be possible
to set the CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine
Caller ID
which CallerID will be used if they exist:

The CallerID configured for the extension will be looked up first.

If no CallerID configured for the extension, the CallerID configured for
the trunk will be used.

If the above two are missing, the "Global Outbound CID" defined in
Web GUI->PBX->Internal Options->General will be used.
Need Registration
Username
Password
Auth ID
Select whether the trunk needs to register on the external server or not
when "Register SIP Trunk" type is selected. The default setting is No.
Enter the username to register to the trunk from the provider when
"Register SIP Trunk" type is selected.
Enter the password to register to the trunk from the provider when
"Register SIP Trunk" is selected.
Enter the Authentication ID for "Register SIP Trunk" type.
Enable automatic recording for the calls using this trunk (for SIP trunk
Auto Record
only). The default setting is disabled. The recording files can be accessed
under web GUI->CDR->Recording Files.
Table 57: SIP Register Trunk Configuration Parameters
Basic Settings
Provider Name
Configure a unique label to identify this trunk when listed in outbound rules,
inbound rules and etc.
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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 "All - UDP Primary".
Transport

UDP Only

TCP Only

TLS Only

All - UDP Primary: UDP is the primary transport protocol when all the
other SIP transport methods are available too.

All - TCP Primary: TCP is the primary transport protocol when all the
other SIP transport methods are available too.

All – TLS Primary: TLS is the primary transport protocol when all the
other SIP transport methods are available too.
Keep the CID from the inbound call when dialing out. This setting will
Keep Original CID
override “Keep Trunk CID” option. Please make sure that the peer PBX at
the other side supports to match user entry using “username” field from
authentication line.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension’s CID when
the extension has CID configured. The default setting is “No”.
Turn on this option when the PBX is using public IP and communicating
NAT
with devices behind NAT. If there is one-way audio issue, usually it’s
related to NAT configuration or SIP/RTP port configuration on the firewall.
If selected, the trunk will be disabled.
Disable This Trunk
Note:
If a current SIP trunk is disabled, UCM6510 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
TEL URI
the Request-Line and TO header in the SIP request to indicate the E.164
number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP
request. The default setting is disabled.
Need Registration
Username
Password
Auth ID
Select whether the trunk needs to register on the external server or not
when "Register SIP Trunk" type is selected. The default setting is No.
Enter the username to register to the trunk from the provider when
"Register SIP Trunk" type is selected.
Enter the password to register to the trunk from the provider when
"Register SIP Trunk" is selected.
Enter the Authentication ID for "Register SIP Trunk" type.
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Enable automatic recording for the calls using this trunk (for SIP trunk
Auto Record
only). The default setting is disabled. The recording files can be accessed
under web GUI->CDR->Recording Files.
Advanced Settings
Select audio and video codec for the VoIP trunk. The available codecs are:
Codec Preference
PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, ILBC,
ADPCM, H.264, H.263, H.263p.
Configure the actual domain name where the extension comes from. This
From Domain
can be used to override the From Header.
For example, "trunk.UCM6510.provider.com" is the From Domain in From
Header: sip:1234567@trunk.UCM6510.provider.com.
Configure the actual user name of the extension. This can be used to
override the From Header. There are cases where there is a single ID for
From User
registration (single trunk) with multiple DIDs.
For
example,
"1234567"
is
the
From
User
in From
Header:
sip:1234567@trunk.UCM6510.provider.com.
If enabled, the SIP INVITE message sent to the trunk will contain PPI (PPreferred-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 is allowed to be contained in the SIP
INVITE message.
If enabled, the SIP INVITE message sent to the trunk will contain PAI (PAsserted-Identity) header. The default setting is “No”.
Send PAI Header
Note:
“Send PPI Header” and “Send PAI Header” cannot be enabled at the same
time. Only one of the two headers is allowed to be contained in the SIP
INVITE message.
Outbound Proxy Support
Outbound Proxy
DID Mode
Select to enable outbound proxy in this trunk. The default setting is "No".
When outbound proxy support is enabled, enter the IP address or URL of
the outbound proxy.
Configure where to get the destination ID of an incoming SIP call, from SIP
Request-line or To-header. The default is set to "Request-line".
Configure the default DTMF mode when sending DTMF on this trunk.
DTMF Mode

Default: The global setting of DTMF mode will be used. The global
setting for DTMF Mode setting is under web UI->PBX->SIP Settings>ToS.
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
RFC2833: Send DTMF using RFC2833.

Info: Send DTMF using SIP INFO message.

Inband: Send DTMF using inband audio. This requires 64-bit codec,
i.e., PCMU and PCMA.

Auto: Send DTMF using RFC2833 if offered. Otherwise, inband will be
used.
Enable Qualify
If enabled, the UCM6510 will regularly send SIP OPTIONS to the device
to check if the device is still online. The default setting is "No".
When "Enable Qualify" option is set to "Yes", configure the timeout (in ms)
Qualify Timeout
for the Qualify SIP message. If no response is received within the timeout,
the device is considered offline. The default setting is 1000ms.
When "Enable Qualify" option is set to "Yes", configure the interval (in
Qualify Frequency
seconds) of the SIP OPTIONS message sent to the device to check if the
device is still online. The default setting is 60 seconds.
Maximum Number of Call
The maximum number of concurrent calls using the trunk. The default
Lines
settings 0, which means no limite.
Select Fax mode. The default setting is “None”.
Fax Mode

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
SRTP
Enable SRTP for the VoIP trunk. The default setting is "No".
CC Settings
Enable CC
If enabled, the system will automatically alert the user when a called party
is available, given that a previous call to that party failed for some reason.
Configure the maximum number of CCSS agents which may be allocated
CC Max Agents
for this channel. In other words, this number serves as the maximum
number of CC requests this channel is allowed to make. The minimum
value is 1.
Configure the maximum number of monitor structures which may be
CC Max Monitors
created for this device. In other words, this number tells how many callers
may request CC services for a specific device at one time. The minimum
value is 1.
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Table 58: SIP Peer Trunk Configuration Parameters
Basic Settings
Provider Name
Host Name
Configure a unique label to identify this trunk when listed in outbound rules,
inbound rules and etc.
Configure the IP address or URL for the VoIP provider’s server of the trunk.
Configure the SIP transport protocol to be used in this trunk. The default
setting is "All - UDP Primary".
Transport

UDP Only

TCP Only

TLS Only

All - UDP Primary: UDP is the primary transport protocol when all the
other SIP transport methods are available too.

All - TCP Primary: TCP is the primary transport protocol when all the
other SIP transport methods are available too.

All – TLS Primary: TLS is the primary transport protocol when all the
other SIP transport methods are available too.
Keep the CID from the inbound call when dialing out, this setting will
Keep Original CID
override “Keep Trunk CID” option. Please make sure that the peer PBX at
the other side supports to match user entry using “username” field from
authentication line.
Keep Trunk CID
If enabled, the trunk CID will not be overridden by extension’s CID when
the extension has CID configured. The default setting is “No”.
Turn on this option when the PBX is using public IP and communicating
NAT
with devices behind NAT. If there is one-way audio issue, usually it’s
related to NAT configuration or SIP/RTP port configuration on the firewall.
If selected, the trunk will be disabled.
Disable This Trunk
Note:
If a current SIP trunk is disabled, UCM6510 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
TEL URI
the Request-Line and TO header in the SIP request to indicate the E.164
number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP
request. The default setting is disabled.
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
Caller ID
to set the CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine
which CallerID will be used if they exist:

The CallerID configured for the extension will be looked up first.
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
If no CallerID configured for the extension, the CallerID configured for
the trunk will be used.

If the above two are missing, the "Global Outbound CID" defined in
Web GUI->PBX->Internal Options->General will be used.
CallerID Name
Configure the name of the caller to be displayed when the extension has
no CallerID Name configured.
Enable automatic recording for the calls using this trunk (for SIP trunk
Auto Record
only). The default setting is disabled. The recording files can be accessed
under web GUI->CDR->Recording Files.
Advanced Settings
Select audio and video codec for the VoIP trunk. The available codecs are:
Codec Preference
PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, ILBC,
ADPCM, H.264, H.263, H.263p.
DID Mode
Configure where to get the destination ID of an incoming SIP call, from SIP
Request-line or To-header. The default is set to "Request-line".
Configure the default DTMF mode when sending DTMF on this trunk.

Default: The global setting of DTMF mode will be used. The global
setting for DTMF Mode setting is under web UI->PBX->SIP Settings>ToS.
DTMF Mode

RFC2833: Send DTMF using RFC2833.

Info: Send DTMF using SIP INFO message.

Inband: Send DTMF using inband audio. This requires 64 bit codec,
i.e., PCMU and PCMA.

Auto: Send DTMF using RFC2833 if offered. Otherwise, inband will be
used.
Enable Qualify
If enabled, the UCM6510 will regularly send SIP OPTIONS to the device
to check if the device is still online. The default setting is "No".
When "Enable Qualify" option is set to "Yes", configure the timeout (in ms)
Qualify Timeout
for the Qualify SIP message. If no response is received within the timeout,
the device is considered offline. The default setting is 1000ms.
When "Enable Qualify" option is set to "Yes", configure the interval (in
Qualify Frequency
seconds) of the SIP OPTIONS message sent to the device to check if the
device is still online. The default setting is 60 seconds.
Maximum Number of Call
The maximum number of concurrent calls using the trunk. The default
Lines
settings 0, which means no limit.
Select Fax mode. The default setting is “None”.
Fax Mode

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
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configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
SRTP
Enable SRTP for the VoIP trunk. The default setting is "No".
If enabled, the local UCM6510 will automatically provide and update the
Sync LDAP Enable
local LDAP contacts to the remote UCM6510 SIP peer trunk. In order to
ensure successful synchronization, the remote UCM6510 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
Sync LDAP Password
during the LDAP sync process. The password must be the same on both
UCM6510 peers to ensure successful synchronization.
Sync LDAP Port
Configure TCP port used LDAP sync feature between two peer UCM6510.
Specify an outbound rule for LDAP sync feature. UCM6510 will
LDAP Outbound Rule
automatically modify the remote contacts by adding prefix parsed from this
rule.
LDAP Dialed Prefix
Specify the prefix for LDAP sync feature. The UCM6510 will automatically
modify the remote contacts by adding this prefix.
CC Settings
Enable CC
If enabled, the system will automatically alert the user when a called party
is available, given that a previous call to that party failed for some reason.
Configure the maximum number of CCSS agents which may be allocated
CC Max Agents
for this channel. In other words, this number serves as the maximum
number of CC requests this channel is allowed to make. The minimum
value is 1.
Configure the maximum number of monitor structures which may be
CC Max Monitors
created for this device. In other words, this number tells how many callers
may request CC services for a specific device at one time. Min. value is 1.
Table 59: Create New IAX Trunk
Select the VoIP trunk type.
Type
Provider Name
Host Name
Keep Trunk CID

Peer IAX Trunk

Register IAX Trunk
Configure a unique label to identify this trunk when listed in outbound rules,
inbound rules and etc.
Configure the IP address or URL for the VoIP provider’s server of the trunk.
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
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Username
Password
Disable This Trunk
Enter the username to register to the trunk from the provider when
"Register IAX Trunk" type is selected.
Enter the password to register to the trunk from the provider when
"Register IAX Trunk" type is selected.
If selected, the trunk will be disabled.
Table 60: IAX Register Trunk Configuration Parameters
Basic Settings
Provider Name
Host Name
Keep Trunk CID
Disable This Trunk
Configure a unique label to identify this trunk when listed in outbound rules,
inbound rules and etc.
Configure the IP address or URL for the VoIP provider’s server of the trunk.
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
If selected, the trunk will be disabled.
Configure the Caller ID. This is the number that the trunk will try to use
when making outbound calls. For some providers, it might not be possible
to set the CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine
Caller ID
which CallerID will be used if they exist:

The CallerID configured for the extension will be looked up first.

If no CallerID configured for the extension, the CallerID configured for
the trunk will be used.

If the above two are missing, the "Global Outbound CID" defined in
Web GUI->PBX->Internal Options->General will be used.
CallerID Name
Configure the name of the caller to be displayed when the extension has
no CallerID Name configured.
Username
Enter the username to register to the trunk from the provider.
Password
Enter the password to register to the trunk from the provider.
Advanced Settings
Select audio and video codec for the VoIP trunk. The available codecs are:
Codec Preference
PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, ILBC,
ADPCM, H.264, H.263, H.263p.
Enable Qualify
If enabled, the UCM6510 will regularly send SIP OPTIONS to the device
to check if the device is still online. The default setting is "No".
When "Enable Qualify" option is set to "Yes", configure the timeout (in ms)
Qualify Timeout
for the Qualify SIP message. If no response is received within the timeout,
the device is considered offline. The default setting is 1000ms.
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When "Enable Qualify" option is set to "Yes", configure the interval (in
Qualify Frequency
seconds) of the SIP OPTIONS message sent to the device to check if the
device is still online. The default setting is 60 seconds.
Maximum Number of Call
The maximum number of concurrent calls using the trunk. The default
Lines
settings 0, 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 the received Fax will be sent to the Email address
Fax Mode
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Table 61: IAX Peer Trunk Configuration Parameters
Basic Settings
Provider Name
Host Name
Keep Trunk CID
Disable This Trunk
Configure a unique label to identify this trunk when listed in outbound rules,
inbound rules and etc.
Configure the IP address or URL for the VoIP provider’s server of the trunk.
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
If selected, the trunk will be disabled.
Configure the Caller ID. This is the number that the trunk will try to use
when making outbound calls. For some providers, it might not be possible
to set the CallerID with this option and this option will be ignored.
When making outgoing calls, the following rules are used to determine
Caller ID
which CallerID will be used if they exist:

The CallerID configured for the extension will be looked up first.

If no CallerID configured for the extension, the CallerID configured for
the trunk will be used.

If the above two are missing, the "Global Outbound CID" defined in
Web GUI->PBX->Internal Options->General will be used.
CallerID Name
Configure the name of the caller to be displayed when the extension has
no CallerID Name configured.
Advanced Settings
Select audio and video codec for the VoIP trunk. The available codecs are:
Codec Preference
PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723, ILBC,
ADPCM, H.264, H.263, H.263p.
Enable Qualify
If enabled, the UCM6510 will regularly send SIP OPTIONS to the device
to check if the device is still online. The default setting is "No".
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When "Enable Qualify" option is set to "Yes", configure the timeout (in ms)
Qualify Timeout
for the Qualify SIP message. If no response is received within the timeout,
the device is considered offline. The default setting is 1000ms.
When "Enable Qualify" option is set to "Yes", configure the interval (in
Qualify Frequency
seconds) of the SIP OPTIONS message sent to the device to check if the
device is still online. The default setting is 60 seconds.
Maximum Number of Call
The maximum number of concurrent calls using the trunk. The default
Lines
settings 0, 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 the received Fax will be sent to the Email address
Fax Mode
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Direct Outward Dialing (DOD) via VoIP Trunks
The UCM6510 provides Direct Outward Dialing (DOD) which is a service of a local phone company (or
local exchange carrier) that allows subscribers within a company's PBX system to connect to outside lines
directly.
Example of how DOD is used:
Company ABC has a SIP trunk. This SIP trunk has 4 DIDs associated to it. The main number of the office
is routed to an auto attendant. The other three numbers are direct lines to specific users of the company.
At the moment when a user makes an outbound call their caller ID shows up as the main office number.
This poses a problem as the CEO would like their calls to come from their direct line. This can be
accomplished by configuring DOD for the CEO’s extension.
Steps on how to configure DOD on the UCM:
1. To setup DOD go to UCM6510 web GUI->PBX->Basic/Call Routes->VoIP Trunks page.
2. Click
to access the DOD options for the selected SIP Trunk.
3. Click "Create a new DOD" to begin your DOD setup.
4. For "DOD Number" enter one of the numbers (DIDs) from your SIP trunk provider. In the example above
Company ABC received 4 DIDs from their provider. ABC will enter in the number for the CEO's direct
line.
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5. Select an extension from the "Available Extensions" list. Users have the option of selecting more than
one extension. In this case, Company ABC would select the CEO's extension. After making the
selection, click on the
button to move the extension(s) to the "Selected Extensions" list.
Figure 106: DOD extension selection
6. Click "Save" at the bottom.
Once completed, the user will return to the Edit DOD page that shows all the extensions that are associated
to a particular DOD.
Figure 107: Edit DOD
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SLA STATION
UCM6510 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 UCM6510 is similar to BLF but SLA is used to monitor external line i.e., analog trunk
on the UCM6510. 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 UCM6510.
Create/Edit SLA Station
SLA Station can be configured on web GUI->PBX->Basic/Call Routes->SLA Station.
Figure 108: SLA Station

Click on “Create New SLA Station” to add a SLA Station.

Click on
to edit the SLA Station. The following table shows the SLA Station configuration
parameters.

Click on
to delete the SLA Station.
Table 62: SLA Station Configuration Parameters
Configure a name to identify the SLA Station.
Station
Specify a SIP extension as a station that will be using SLA.
Available SLA Trunks
Existing Analog Trunks with SLA Mode enabled will be listed here.
Select a trunk for this SLA from the Available SLA Trunks list. Click on
Selected SLA Trunks
to arrange the order. If there are multiple trunks selected, when
there are calls on those trunks at the same time, pressing the LINE key on
the phone will pick up the call on the first trunk here.
SLA Station Options
Ring Timeout
Configure the time (in seconds) to ring the station before the call is
considered unanswered. No timeout is set by default. If set to 0, there will
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be no timeout.
Configure the time (in seconds) for delay before ringing the station when a
Ring Delay
call first coming in on the shared line. No delay is set by default. If set to 0,
there will be no delay.
This option defines the competence of the hold action for one particular
Hold Access
trunk. If set to “open”, any station could hold a call on that trunk or resume
one held session; if set to “private”, only the station that places the trunk
call on hold could resume the session. The default setting is “open”.
Sample Configuration
1. On the UCM6510, go to web UI->Basic/Call Routes->Analog Trunks page. Create analog trunk or
edit the existing analog trunk. Make sure “SLA Mode” is enabled for the analog trunk. Once enabled,
this analog trunk will be only available for the SLA stations created under web UI->Basic/Call Routes>SLA Station page.
Figure 109: Enable SLA Mode for Analog Trunk
Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.
Figure 110: Analog Trunk with SLA Mode Enabled
2. On the UCM6510, go to web UI->Basic/Call Routes->SLA Station page, click on “Create New SLA
Station”. Please refer to section [Create/Edit SLA Station] for the configuration parameters. Users can
create one or more SLA stations to monitor the analog trunk. The following figure shows two stations,
1002 and 1005, are configured to be associated with SLA trunk “fxo1”.
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Figure 111: SLA Example - SLA Station

On the SIP phone 1, configure to register UCM6510 extension 1002. Configure the MPK as BLF mode
and the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case.

On the SIP phone 2, configure to register UCM6510 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
Outbound Routes
In the UCM6510, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern.
This allows different patterns to be dialed through different trunks (e.g., "Local" 7-digit dials through a FXO
while "Long distance" 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to
be used when the primary trunk fails.
Go to web GUI->PBX->Basic/Call Routes->Outbound Routes to add and edit outbound rules.

Click on "Create New Outbound Rule" to add a new outbound route.

Click on
to edit the outbound route.

Click on
to delete the outbound route.

On the UCM6510, the outbound route priority is based on “Best matching pattern”. For example, the
UCM6510 has outbound route A with pattern 1xxx and outbound route B with pattern 10xx configured.
When dialing 1000 for outbound call, outbound route B will always be used first. This is because pattern
10xx is a better match than pattern 1xxx. Only when there are multiple outbound routes with the same
pattern configured, users can click on
to move the outbound route up/down to arrange
the priority among those outbound routes.
Table 63: Outbound Route Configuration Parameters
Calling Rule Name
Configure the name of the calling rule (e.g., local, long_distance, and etc).
Letters, digits, _ and - are allowed.

All patterns are prefixed with the "_".

Special characters:
X: Any Digit from 0-9.
Pattern
Z: Any Digit from 1-9.
N: Any Digit from 2-9.
".": Wildcard. Match one or more characters.
"!": Wildcard. Match zero or more characters immediately.
Example: [12345-9] - Any digit from 1 to 9.
Password
Configure the password for users to use this rule when making outbound
calls.
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Call Duration Limit
Once call duration limit is enabled, it will set the maximum duration of callblocking.
User can customize the maximum call duration (in seconds) that is allowed
The 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.
Warning Repeat Interval
Once this option is set to ‘z’, it will repeatedly be warning caller every z
seconds after the first warning.
Select privilege level for the outbound rule.
Internal: The lowest level required. All users can use this rule.

Local: Users with Local, National, or International level are allowed to
use this rule.

National: Users with National or International level are allowed to use
this rule.
Privilege Level

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.
3. Select available extensions/extension groups from the left to the right.
This allows users to specify arbitrary single extensions available in the
PBX.
Enable Filter on Source
Caller ID
4. Custom Dynamic Route: define the pattern for the source caller ID.
This allows users to define extension range instead of selecting them
one by one.

All patterns are prefixed with the "_".

Special characters:
X: Any Digit from 0-9.
Z: Any Digit from 1-9.
N: Any Digit from 2-9.
".": Wildcard. Match one or more characters.
"!": Wildcard. Match zero or more characters immediately.
Example: [12345-9] - Any digit from 1 to 9.
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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.
Strip
Example:
The users will dial 9 as the first digit of a long distance calls. However, 9
should not be sent out via analog lines and the PSTN line. In this case, 1
digit should be stripped before the call is placed.
Prepend
Specify the digits to be prepended before the call is placed via the trunk.
Those digits will be prepended after the dialing number is stripped.
Use Failover Trunk
Failover trunks can be used to make sure that a call goes through an
alternate route, when the primary trunk is busy or down. If "Use Failover
Trunk" is enabled and "Failover trunk" is defined, the calls that cannot be
placed via the regular trunk may have a secondary trunk to go through.
Failover Trunk
Example:
The user's primary trunk is a VoIP trunk and the user would like to use the
PSTN when the VoIP trunk is not available. The PSTN trunk can be
configured as the failover trunk of the VoIP trunk.
Allows the user to specify the number of digits that will be stripped from the
beginning of the dialed string before the call is placed via the selected trunk.
Strip
Example:
The users will dial 9 as the first digit of a long distance calls. However, 9
should not be sent out via analog lines and the PSTN line. In this case, 1
digit should be stripped before the call is placed.
Prepend
Specify the digits to be prepended before the call is placed via the trunk.
Those digits will be prepended after the dialing number is stripped.
Country Codes
The UCM6510 allows users to put country code restrictions on specific outbound routes. Once the restriction is
enabled, call to the restricted country cannot be placed on that specific trunk. To configure this feature, please
navigate to web UI-> PBX -> Basic -> Outbound Routes -> Country Codes.
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Figure 113: Country Codes
Inbound Routes
Inbound routes can be configured via web GUI->PBX->Basic/Call Routes->Inbound Routes.

Click on "Create New Inbound Rule" button to add a new inbound route.

Click on "Blacklist" button 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 64: 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.
DID Pattern
".": Wildcard. Match one or more characters.
"!": Wildcard. Match zero or more characters immediately.
Example: [12345-9] - Any digit from 1 to 9.

The pattern can be composed of two parts, divided by a ‘/’ character.
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The first part is used to specify the dialed number the second part is
used to specify the caller ID and it is optional, if set it means only the
extension with the specific caller ID is allowed to call in or call out.
For
example, patter '_2XXX/1234' means the only extension with the caller
ID '1234' is allowed to use this rule.
Prepend Trunk Name
Alert-Info
If enabled, the trunk name will be added to the caller id name as the
displayed caller id name.
Configure the Alert-Info, when UCM6510 receives an INVITE request, the
Alert-Info header field specifies an alternative ring tone to the UAS.
Multiple mode allows user to switch between destinations of the inbound
Inbound Multiple Mode
rule by feature codes. Configure related feature codes in the “Feature
Codes” page. If this option is enabled, user can use feature code to switch
between different destinations.
Select the default destination for the inbound call.
Default Destination

Extension

Voicemail

Conference Room

Call Queue

Ring Group

Paging/Intercom

Voicemail Group

Fax

DISA

IVR

External Number

By DID
When "By DID" is used, the UCM6510 will look for the destination based
on the number dialed, which could be local extensions, conference, call
queue, ring group, paging/intercom group, IVR, voicemail groups and
Fax extension as configured in "DID destination". If the dialed number
matches the DID pattern, the call will be allowed to go through.
Strip
Prepend

Dial By Name

Callback
Specify the number of digits to strip from the beginning of the DID. This is
used when "By DID" is selected in "Default Destination".
Specify the digits to be prepended before the call is placed via the trunk.
Those digits will be prepended after the dialing number is stripped.
Configure to allow the inbound call to dial out from the PBX's trunk or not.
Dial Trunk
The default setting is disabled. Please be aware of potential security risk if
"Dial Trunk" is enabled. The inbound call might be able to dial out
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international calls from the PBX's trunk if allowed by the privilege level.
Select the DID destination if "By DID" is selected in "Default Destination".
Only the selected category can be reached by DID using this inbound route.
DID Destination

Extension

Conference

Call Queue

Ring Group

Paging/Intercom Group

IVR

Voicemail Groups

Fax Extension

Dial By Name

All
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.
Date
Week
Select "By Week" or "By Day" and specify the date for the trunk to use the
inbound rule.
Select the day in the week to use the inbound rule.
Select the destination for the inbound call under the defined time condition.
Destination

Extension

Voicemail

Conference Room

Call Queue

Ring Group

Paging/Intercom

Voicemail Group

Fax

DISA

IVR

By DID
When "By DID" is used, the UCM6510 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
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Inbound Route: Prepend Example
UCM6510 now allows user to prepend digits to an inbound DID pattern, with strip taking precedence over
prepend. With the ability to prepend digits in inbound route DID pattern, user no longer needs to create
multiple routes for the same trunk in order to route calls to different extensions.
Figure 114: Inbound Route feature: Prepend
The following example demonstrates the process,
1. If Trunk provides a DID pattern of 18005251163.
2. If Strip is set to 8, UCM6510 will strip the first 8 digits.
3. If Prepend is set to 2, UCM6510 will then prepend a 2 to the stripped number, now the number
become 2163.
4. UCM6510 will now forward the incoming call to extension 2163.
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Inbound Route: Multiple Mode
In UCM6510, 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.
Figure 115: Inbound Route - Multiple Mode
When Multiple Mode is enabled for the inbound route, the user can configure a “Default Destination” and a “Mode
1” destination for this route. By default, the call coming into this inbound route will be routed to the default
destination.
SIP end devices that have registered on the UCM6510 can dial feature code *62 to switch to inbound route “Mode
1” and dial feature code *61 to switch back to “Default Destination”. Switching between different mode can be
easily done without web UI login.
For example, the customer service hotline destination has to be set to a different IVR after 7PM. The user can
dial *62 to switch to “Mode 1” with that IVR set as the destination before off work.
Fax Intelligent Route
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The UCM6510 can automatically detect Fax and phone signal coming from the FXO port, and then forward Fax
or phone signal to the right destination. For example, when a regular phone call is coming, the UCM65100 will be
able to detect the phone signal and forward it through the correct inbound route to the destination; if Fax signal is
coming, the UCM6510 will be able to forward it to the FXS extension where the Fax machine is connected.
Fax with Two Media
UCM6510 with Asterisk 13 system now supports fax re-invite with multiple codec negotiation. If a fax re-invite
contains both T.38 and PCMA/PCMU codec, UCM6510 will choose T.38 codec over PCMA/PCMU. Where in the
old Asterisk 1.8 system, multiple codec in the re-invite is prohibited and will be dropped by UCM6510.
Blacklist Configurations
In the UCM6510, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature and
manage the Blacklist by clicking on "Blacklist".

Select the checkbox for "Blacklist Enable" to turn on Blacklist feature for all inbound routes. Blacklist is
disabled by default.

Enter a number in "Add Blacklist Number" field and then click
to add to the list. Anonymous can
also be added as a Blacklist Number.

To remove a number from the Blacklist, select the number in "Blacklist list" and click on
.
Figure 116: Blacklist Configuration Parameters
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
To add blacklist number in batch, click on
to upload blacklist file in csv format. The supported csv
format is as below.
Figure 117: Blacklist csv File
--------------------------------------------------------------------------------------------------------------------- ----------------------Note:
Users could also add a number to the Blacklist or remove a number from the Blacklist by dialing the feature
code for "Blacklist Add' (default: *40) and "Blacklist Remove" (default: *41) from an extension. The feature
code can be configured under web GUI->PBX->Internal Options->Feature Codes.
--------------------------------------------------------------------------------------------------------------------------------------------
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CONFERENCE BRIDGE
The UCM6510 supports Conference Bridge allowing 64 participants with up to 8 bridges at the same time.
The conference bridge configurations can be accessed under web GUI->PBX->Call Features>Conference. In this page, users could create, edit, view, invite, manage the participants and delete
conference bridges. The conference bridge status and conference call recordings (if recording is enabled)
will be displayed in this web page as well.
Conference Bridge Configurations

Click on "Create New Conference Room" to add a new conference bridge.

Click on
to edit the conference bridge.

Click on
to delete the conference bridge.
Table 65: Conference Bridge Configuration Parameters
Extension
Configure the conference number for the users to dial into the conference.
When configured, the users who would like to join the conference call must
enter this password before accessing the conference bridge.
Password
Note:

If "Public Mode" is enabled, the password is not required to join the
conference bridge thus this field is invalid.

The password has to be at least 4 characters.
Configure the password to join the conference bridge as administrator.
Conference administrator can manage the conference call via IVR (if
"Enable Caller Menu" is enabled) as well as invite other parties to join the
conference by dialing "0" (permission required from the invited party) or "1"
Admin Password
(permission not required from the invited party) during the conference call.
Note:

If "Public Mode" is enabled, the password is not required to join the
conference bridge thus this field is invalid.

Enable Caller Menu
The password has to be at least 4 characters.
If enabled, conference participant could press the * key to access the
conference bridge menu. The default setting is "No".
If enabled, the calls in this conference bridge will be recorded automatically
Record Conference
in a .wav format file. All the recording files will be displayed and can be
downloaded in the conference web page. The default setting is "No".
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If enabled, if there are users joining or leaving the conference, voice prompt
Quiet Mode
or notification tone won't be played. The default setting is "No".
Note:
"Quiet Mode" and "Announce Callers" cannot be enabled at the same time.
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)
Enable User Invite
to join the conference. The default setting is "No".
Note:
Conference administrator can always invite other users without enabling
this option.
If enabled, the caller will be announced to all conference participants when
Announce Callers
there the caller joins the conference. The default setting is "No".
Note:
"Quiet Mode" and "Announce Callers" cannot be enabled at the same time.
Public Mode
Play Hold Music
If enabled, no authentication will be required when joining the conference
call. The default setting is "Yes".
If enabled, the UCM6510 will play Hold music to the first participant in the
conference until another user joins in. The default setting is "No".
Select the music on hold class to be played in conference call. This option
Music On Hold
shows up if "Play Hold Music For First Caller" is enabled. Music On Hold
class can be set up under web UI->PBX->Internal Options->Music On
Hold.
Skip Authentication When
If enabled, the invitation from web GUI for a conference bridge with
Inviting User via Trunk from
password will skip the authentication for the invited users. The default
web GUI
setting is "No".
Join a Conference Call
Users could dial the conference bridge extension to join the conference. If password is required, enter the
password to join the conference as a normal user, or enter the admin password to join the conference as
administrator.
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Invite Other Parties to Join Conference
When using the UCM6510 conference bridge, there are two ways to invite other parties to join the
conference.

Invite from web GUI
For each conference bridge in UCM6510 web GUI->PBX->Call Features->Conference, there is an icon
for option "Invite a participant". Click on it and enter the number of the party you would like to invite.
Then click on "Add". A call will be sent to this number to join it into the conference.
Figure 118: Conference Invitation From web GUI

Invite by dialing 0 or 1 during conference call
A conference participant can invite other parties to the conference by dialing from the phone during the
conference call. Please make sure option "Enable User Invite" is turned on for the conference bridge
first. Enter 0 or 1 during the conference call. Follow the voice prompt to input the number of the party
you would like to invite. A call will be sent to this number to join it into the conference.
0: If 0 is entered to invite other party, once the invited party picks up the invitation call, a permission will
be asked to "accept" or "reject" the invitation before joining the conference.
1: If 1 is entered to invite other party, no permission will be required from the invited party.
Note:
Conference administrator can always invite other parties from the phone during the call by entering 0 or 1.
To join a conference bridge as administrator, enter the admin password when joining the conference. A
conference bridge can have multiple administrators.
During The Conference
During the conference call, users can manage the conference from web GUI or IVR.
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
Manage the conference call from web GUI.
Log in UCM6510 web GUI during the conference call, the participants in each conference bridge will
be listed.

1. Click on
to kick a participant from the conference.
2. Click on
to mute the participant.
3. Click on
to lock this conference bridge so that other users cannot join it anymore.
4. Click on
to invite other users into the conference bridge.
Manage the conference call from IVR.
If "Enable Caller Menu" is enabled, conference participant can input * to enter the IVR menu for the
conference. Please see options listed in the table below.
Table 66: Conference Caller IVR Menu
Conference Administrator IVR Menu
1
Mute/unmute yourself.
2
Lock/unlock the conference bridge.
3
Kick the last joined user from the conference.
4
Decrease the volume of the conference call.
6
Increase the volume of the conference call.
7
Decrease your volume.
9
Increase your volume.
More options:
8

1: List all users currently in the conference call.

2: Kick all non-Administrator participants from the conference call.

3: Mute/Unmute all non-Administrator participants from the conference call.

4: Enable/disable conference call recording.

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.
6
Increase the volume of the conference call.
7
Decrease your volume.
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9
Increase your volume.
8
Exit the caller menu and return to the conference.
Note:
When there is participant in the conference, the conference bridge configuration cannot be modified.
Record Conference
The UCM6510 allows users to record the conference call and retrieve the recording from web GUI->PBX>Call Features->Conference.
To record the conference call, when the conference bridge is in idle, enable "Record Conference" from the
conference bridge configuration dialog. Save the setting and apply the change. When the conference call
starts, the call will be automatically recorded in .wav format.
The recording files will be listed as below once available. Users could click on
recording or click on
to download the
to delete the recording.
Figure 119: Conference Recording
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CONFERENCE SCHEDULE
Conference Schedule Configuration
Conference Schedule can be found under UCM6510 web UI->PBX->Call Features->Conference Schedule.
Users can create, edit, view and delete a Conference Schedule.

Click on “Create New Conference Schedule” to add a new Conference Schedule.

Click on the scheduled conference to edit or delete the event.
After the user configures UCM6510 with Google Service Settings [Google Service Settings Support] and
enables Google Calendar for Conference Schedule, the conference schedule on the UCM6510 can be
synchronized with Google Calendar for authorized Google account.
Table 67: Conference Schedule Parameters
Schedule Options
Conference Topic
Conference Room
Configure the name of the scheduled conference. Letters, digits, _ and are allowed.
Select a conference room for this scheduled conference.
Set kick time before conference starts. When kick time is reached, a
warning prompt will be played for all attendees in the conference room.
Kick Time(m)
After 5 minutes, this conference room will be cleared and locked for the
scheduled conference to begin.
Note: Kick Time cannot be less than 6 minutes in order to clear the
conference room.
Description
The description of scheduled conference.
Repeat
Repeat interval of scheduled conference. By default it’s set to single event.
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.
Select this option to sync scheduled conference with Google Calendar.
Enable Google Calendar
Note: Google Service Setting OAuth2.0 must be configured on the
UCM6510. Please refer to section [Google Service Settings Support].
Select the administrator of scheduled conference from selected
Conference Administrator
extensions.
Note: “Public Mode” must be disabled from Conference Room Options
tab.
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Local Extension
Select available extensions from the list to attend scheduled conference.
Select available extensions from the remote peer PBX.
Remote Extension
Note: “LDAP Sync” must be enabled on the UCM6510 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 match the pattern on the
outbound route.
Remote Conference
Invite a remote conference.
Conference Room Options
Password
Admin Password
Enable Caller Menu
Record Conference
Configure conference room password. Please note that if “Public Mode” is
enabled, this option is automatically disabled.
Configure the password to join as conference administrator. Please note
that if “Public Mode” is enabled, this option is automatically disabled.
If this option is enabled, conference participants will be able to access
conference bridge menu by pressing the * key.
If this option is enabled, conference call will be recorded in .wav format.
The recorded file can be found from Conference page.
If this option is enabled, the notification tone or voice prompt for joining or
Quiet Mode
leaving the conference won’t be played.
Note: Option “Quiet Mode” and option “Announce Caller” cannot be
enabled at the same time.
If this option is enabled, the participants in the conference won’t be able to
Wait For Admin
hear each other until conference administrator joins the conference.
Note: If “Quiet Mode” is enabled, voice prompt for this option won’t be
played.
If this option is enabled, the user can:

Press ‘0’ to invite others to join the conference with invited party’s
permission
Enable User Invite

Press ‘1’ to invite without invited party’s permission

Press ‘2’ to create a multi-conference bridge to another conference
room

Press ‘3’ to drop all current multi-conference bridges
Note: Conference Administrator is always allowed to access this menu.
If this option is enabled, when a participant joins the conference room,
Announce Callers
participant’s name will be announced to all members in the conference
room.
Note: Option “Quiet Mode” and option “Announce Caller” cannot be
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enabled at the same time.
If this option is enabled, no authentication is required for entering the
Public Mode
conference room.
Note: Please be aware of the potential security risks when turning on this
option.
Play Hold Music
Skip Authentication When
Inviting Users via Trunk from
Web GUI

If this option is enabled, UCM6510 will play Hold Music while there is only
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
password won’t require authentication.
Note: Please be aware of the potential security risks when turning on this
option.
Cleaner Options
Cleaner Options
Enable Conference
If this option is enabled, conference schedules will be automatically
Schedules Cleaner
cleaned as configured.
Conference Schedules Clean
Time
Clean Interval

Enter the clean time (in hours). The valid range is from 0 to 23.
Enter the clean interval (in days). The valid range is from 1 to 30.
Show/hide Conference Schedule Table
Enable this option will allow web UI to display scheduled conference in Conference Schedule Table. Please
see figure below.
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Figure 120: 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,
UCM6510 will send INVITE to the extensions that have been selected for conference.
--------------------------------------------------------------------------------------------------------------------- ----------------------Note:

Please make sure that outbound route is properly configured for remote extensions to join the
conference.

Once Kick Time is reached, Conference Schedule is locked and cannot be modified.
------------------------------------------------------------------------------------------------------------------- -------------------------
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IVR
Configure IVR
IVR configurations can be accessed under the UCM6510 web GUI->PBX->Call Features->IVR. Users
could create, edit, view and delete an IVR.

Click on "Create New IVR" to add a new IVR.

Click on
to edit the IVR configuration.

Click on
to delete the IVR.
Table 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.
This option shows up only when "By DID" is selected. This controls the
destination that can be reached by the external caller via the inbound
route. The DID destination are:
DID Destination

Extension

Conference

Call Queue

Ring Group

Paging/Intercom Groups

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".
Permission
If the user tries to dial outbound calls after dialing into the IVR, the
UCM6510 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.
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Alert-Info
When present in an INVITE request, the alert-Info header field specifies
and alternative ring tone to the UAS.
Select an audio file to play as the welcome prompt for the IVR. Click on
Welcome Prompt
"Prompt" to add additional audio file under web GUI->Internal Options>IVR Prompt.
Configure the timeout between digit entries. After the user enters a digit,
Digit Timeout
the user needs to enter the next digit within the timeout. If no digit is
detected within the timeout, the UCM6510 will consider the entries
complete. The default timeout is 3 seconds.
After playing the prompts in the IVR, the UCM6510 will wait for the DTMF
Response Timeout
entry within the timeout (in seconds). If no DTMF entry is detected within
the timeout, a timeout prompt will be played. The default setting is 10
seconds.
Response Timeout Prompt
Invalid Prompt
Response Timeout Repeat
Loops
Select the prompt message to be played when timeout occurs.
Select the prompt message to be played when an invalid extension is
pressed.
Configure the number of times to repeat the prompt if no DTMF input is
detected. When the loop ends, it will go to the timeout destination if
configured, or hang up. The default setting is 3.
Configure the number of times to repeat the prompt if the DTMF input is
Invalid Repeat Loops
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->Internal Options->Language. The dropdown list shows all
Language
the current available voice prompt languages on the UCM6510. To add
more languages in the list, please download voice prompt package by
selecting "Check Prompt List" under web GUI->PBX->Internal Options>Language.
Key Pressing Events
Key Press Event:
Select the event for each key pressing for 0-9, *, Timeout and Invalid. The
Press 0
event options are:
Press 1
7. Extension
Press 2
8. Voicemail
Press 3
9. Conference Rooms
Press 4
10. Voicemail Group
Press 5
11. IVR
Press 6
12. Ring Group
Press 7
13. Queues
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Press 8
14. Page Group
Press 9
15. Fax
Press *
16. Custom Prompt
Timeout
17. Hangup
Invalid
18. DISA
19. Dial By Name
20. External Number
21. Callback
Create IVR 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 IVR Prompt page. Or users could go to web GUI->PBX>Internal Options->Custom Prompt page directly.
Figure 121: Click On Prompt To Create IVR Prompt
Once the IVR prompt file is successfully added to the UCM6510, it will be added into the prompt list options
for users to select in different IVR scenarios.
Record New IVR Prompt
In the UCM6510 web GUI->PBX->Internal Options->Custom Prompt page, click on "Record New IVR
Prompt" and follow the steps below to record new IVR prompt.
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Figure 122: Record New IVR Prompt

Specify the IVR file name.

Select the format (GSM or WAV) for the IVR prompt file to be recorded.

Select the extension to receive the call from the UCM6510 to record the IVR prompt.

Click the "Record" button. A request will be sent to the UCM6510. The UCM6510 will then call the
extension for recording the IVR prompt from the phone.

Pick up the call from the extension and start the recording following the voice prompt.

The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play or
delete the recording.
Upload IVR Prompt
If the user has a pre-recorded IVR prompt file, click on "Upload IVR Prompt" in web GUI->PBX->Internal
Options->IVR Prompt page to upload the file to the UCM6510. The following are required for the IVR
prompt file to be successfully uploaded and used by the UCM6510:

PCM encoded.

16 bits.

8000Hz mono.

In .mp3 or .wav format; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix.

File size under 5M.
Figure 123: Upload IVR Prompt
Click on
to select audio file from local PC and click on
to start uploading. Once uploaded, the file
will appear in the IVR Prompt web page.
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LANGUAGE SETTINGS FOR VOICE PROMPT
The UCM6510 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
Swedish
Turkish
English (United States) and Chinese voice prompts are built in with the UCM6510 already. The other
languages provided by Grandstream can be downloaded and installed from the UCM6510 web GUI directly.
Additionally, users could customize their own voice prompts, package them and upload to the UCM6510.
Language settings for voice prompt can be accessed under web GUI->PBX->Internal Options>Language.
Download and Install Voice Prompt Package
To download and install voice prompt package in different languages from UCM6510 web GUI, click on
"Check Prompt List" button.
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Figure 124: Language Settings for Voice Prompt
A new dialog window of voice prompt package list will be displayed. Users can see the version number
(latest version available V.S. current installed version), package size and options to upgrade or download
the language.
Figure 125: Voice Prompt Package List
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Click on
to download the language to the UCM6510. The installation will be automatically started once
the downloading is finished.
Figure 126: New Voice Prompt Language Added
A new language option will be displayed after successfully installed. Users then could select it to apply in
the UCM6510 system voice prompt or delete it from the UCM6510.
Customize Specific Prompt
On the UCM6510, if the user needs to replace some specific customized prompt, the user can upload a single
specific customized prompt from web UI->PBX->Internal Options->Language instead of the entire language
pack.
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Figure 127: Upload Single Voice Prompt for Entire Language Pack
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VOICEMAIL
Configure Voicemail
If the voicemail is enabled for UCM6510 extensions, the configurations of the voicemail can be globally set
up and managed under web GUI->PBX->Call Features->Voicemail.
Table 69: Voicemail Settings
Max Greeting
Configure the maximum number of seconds for the voicemail greeting. The
default setting is 60 seconds.
If enabled, the caller can press 0 to exit the voicemail application and
Dial '0' For Operator
connect to the configured operator's extension. The operator extension can
be configured under web GUI->PBX->Internal Options->General.
Max Messages Per Folder
Configure the maximum number of messages per folder in users' voicemail.
The valid range 10 to 1000. The default setting is 50.
Select the maximum duration of the voicemail message. The message will
not be recorded if the duration exceeds the max message time. The default
setting is 15 minutes. The available options are:
Max Message Time

1 minute

2 minutes

5 minutes

15 minutes

30 minutes

Unlimited
Configure the minimum effective duration (in seconds) of a voicemail
message. Messages will be automatically deleted if the duration is shorter
than the Min Effective Message Time. The default setting is 3 seconds. The
available options are:
Min Effective Message Time

No minimum

1 second

2 seconds

3 seconds

4 seconds

5 seconds
Note:
Silence and noise duration are not counted in message time.
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If enabled, the caller ID of the user who has left the message will be
Announce Message Caller-ID
announced at the beginning of the voicemail message. The default setting
is "No".
Announce Message Duration
If enabled, the message duration will be announced at the beginning of the
voicemail message. The default setting is "No".
If enabled, a brief introduction (received time, received from, and etc) of
each message will be played when accessed from the voicemail
Play Envelope
application. The default setting is "Yes".
If enabled, UCM6510 will play from the voice message left most recently; if
Play from Last
disabled, UCM6510 will play from the earliest left voice message
Allow User Review
If enabled, users can review the message following the IVR before sending
the message out. The default setting is "No".
Access Voicemail
If the voicemail is enabled for UCM6510 extensions, the users can dial the voicemail access feature code
(by default *98 or *97) to access the extension’s voicemail. The users will be prompt to enter the voicemail
password and then can enter digits from the phone keypad to navigate in the IVR menu for different options.
Table 70: Voicemail IVR Menu
Main Menu
Sub Menu 1
Sub Menu 2
1 - Send a reply
2 - Call the person who sent this message
3 - Advanced options
3 - Hear the message envelop
4 - Leave a message
1 - New
messages
* - Return to the main menu
5 - Repeat the current message
7 - Delete this message
8 - Forward the message to another user
9 – Save
* - Help
# - Exit
0 - New messages
2 - Change
folders
1 - Old messages
2 - Work messages
3 - Family messages
4 - Friend messages
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# - Cancel
1 - Send a reply
3-
2 - Call the person who sent this message
Advanced
3 - Hear the message envelop
options
4 - Leave a message
* - Return to
the main menu
1 - Accept this recording
1 - Record your unavailable message
2 - Listen to it
3 - Re-record your message
1 - Accept this recording
2 - Record your busy message
2 - Listen to it
3 - Re-record your message
1 - Accept this recording
0 - Mailbox
options
3 - Record your name
2 - Listen to it
3 - Re-record your message
1 - Accept this recording
4 - Record temporary greeting
2 - Listen to it
3 - Re-record your message
5 - Change your password
* - Return to the main menu
Voicemail Email Settings
The UCM6510 can be configured to send the voicemail as attachment to Email. Click on "Voicemail Email
Settings" button to configure the Email attributes and content.
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Figure 128: Voicemail Email Settings
Click on "Load Default Settings" button to view the default template as an example.
Table 71: Voicemail Email Settings
Attach Recordings to E-Mail
Keep Recordings
If enabled, voicemails will be sent to user's Email address. The default
setting is "Yes".
If enabled, voicemail will be stored in the UCM6510 after email is sent. The
default setting is “Yes”.
Fill in the "Subject:" and "Message:" content, to be used in the Email when
sending to the user.
The template variables are:
Template For Voicemail
Emails

\t: TAB

${VM_NAME}: Recipient's first name and last name

${VM_DUR}: The duration of the voicemail message

${VM_MAILBOX}: The recipient's extension

${VM_CALLERID}: The caller ID of the person who has left the
message

${VM_MSGNUM}: The number of messages in the mailbox

${VM_DATE}: The date and time when the message is left
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Configure Voicemail Group
The UCM6510 supports voicemail group and all the extensions added in the group will receive the voicemail
to the group extension. The voicemail group can be configured under web GUI->PBX->Call Features>Voicemail Group. Click on "Create New Voicemail Group" to configure the group.
Figure 129: Voicemail Group
Table 72: Voicemail Group Settings
Voicemail Group Extension
Name
Enter the Voicemail Group Extension. The voicemail messages left to this
extension will be forwarded to all the voicemail group members.
Configure the Name to identify the voicemail group. Letters, digits, _ and are allowed.
Voicemail Password
The Voicemail password for the user to check Voicemail messages.
Email Address
The Email address of current user.
Select available mailboxes from the left list and add them to the right list.
Voicemail Group Mailboxes
The extensions need to have voicemail enabled to be listed in available
mailboxes list.
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RING GROUP
The UCM6510 supports ring group feature with different ring strategies applied to the ring group members.
This section describes the ring group configuration on the UCM6510.
Configure Ring Group
Ring group settings can be accessed via web GUI->PBX->Call Features->Ring Group.
Figure 130: Ring Group

Click on "Create New Ring Group" to add ring group.

Click on
to edit the ring group. The following table shows the ring group configuration parameters.

Click on
to delete the ring group.
Table 73: Ring Group Parameters
Ring Group Name
Extension
Ring Group Members
Configure ring group name to identify the ring group. Letters, digits, _ and are allowed.
Configure the ring group extension.
Select available users from the left side to the ring group member list on the
right side. Click on
to arrange the order.
Select available remote users from the left side to the ring group member
Selected LDAP Numbers
list 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 all the members at the same time when there is incoming call to
the ring group extension. If any of the member answers the call, it will
stop ringing.
Ring Strategy

Ring in order
Ring the members with the order configured in ring group list. If the first
member doesn't answer the call, it will stop ringing the first member and
start ringing the second member.
Music On Hold
Select the “Music On Hold” Class of this Ring Group, “Music On Hold” can
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be managed from the “Music On Hold” panel on the left.
This option allows user to customize prompt for a ring group to announce
Custom Prompt
to caller. The file can be uploaded from web GUI->Internal Options>Custom Prompt, or click the ‘Prompt’ to add additional record.
Configure the number of seconds to ring each member. If set to 0, it will
Ring Timeout on Each
Member
keep ringing. The default setting is 30 seconds.
Note:
The actual ring timeout might be overridden by users if the phone has ring
timeout settings as well.
Auto Record
Once this option is enabled, the calls using this extension or trunk will be
automatically recorded.
If enabled, users could select extension, voicemail, ring group, IVR, call
Enable Destination
queue, voicemail group as the destination if the call to the ring group has
no answer. Secret and Email address are required if voicemail is selected
as the destination.
Configure the password to access the ring group extension's voicemail.
Secret
Note:
The password has to be at least 4 characters.
Configure the Email address of the ring group extension's voicemail. If
Email Address
"Attach Recordings to E-mail" is enabled from web GUI->PBX->Voicemail>Voicemail Email Settings, the voicemail can be sent to the ring group's
Email address as attachment.
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Figure 131: Ring Group Configuration
Remote Extension in Ring Group
Remote extensions from the peer trunk of a remote UCM6510 can be included in the ring group with local
extension. An example of Ring Group with peer extensions is presented in the following:
1. Creating SIP Peer Trunk between both UCM6510_A and UCM6510_B. SIP Trunk can be found
under web UI-> PBX-> Basic/Call Routes-> VoIP Trunks. Also, please configure their
Inbound/Outbound routes accordingly.
2. Click edit button in the menu
, and check if Sync LDAP Enable is selected, this
option will allow UCM6510_A update remote LDAP server automatically from peer UCM6510_B. In
addition, Sync LDAP Password must match for UCM6510_A and UCM6510_B in order to sync
LDAP contact automatically. Port number can be anything between 0~65535, and use the outbound
rule created in step 1 for the LDAP Outbound Rule option.
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Figure 132: Sync LDAP Server option
3. In case if LDAP server doesn’t sync automatically, user can manually sync LDAP server. Under VoIP
Trunks page, click sync button shown in the following figure to manually sync LDAP contacts from
peer UCM6510.
Figure 133: Manually Sync LDAP Server
4. Under Ring Groups setting page, click
. Ring Groups can be found
under web UI-> PBX-> Call Features-> Ring Groups.
5. If LDAP server is synced correctly, Available LDAP Numbers box will display available remote
extensions that can be included in the current ring group. Please also make sure the extensions in
the peer UCM6510 can be included into that UCM6510’s LDAP contact.
Figure 134: Ring Group Remote Extension
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PAGING AND INTERCOM GROUP
Paging and Intercom Group can be used to make an announcement over the speaker on a group of phones.
Targeted phones will not ring but answer immediately using speaker. The UCM6510 paging and intercom
can be used via feature code to a single extension or a paging/intercom group. This sections describes the
configuration of paging/intercom group under web GUI->PBX->Call Features->Paging/Intercom.
Configure Paging/Intercom Group

Click on "Create New Paging/Intercom Group" to add paging/intercom group.
Figure 135: Paging/Intercom Group
Table 74: Paging/Intercom Group Configuration Parameters
Name
Configure paging/intercom group name.
Extension
Configure the paging/intercom group extension.
Type
Select "2-way Intercom" or "1-way Page".
This option is setting a custom prompt used as announcement placed to
the person receiving a paging/internal call. The file can be uploaded from
Custom Prompt
the web GUI->Internal Options->Custom Prompt, or click the ‘Prompt’ to
add additional recorded.
Page/Intercom Group
Select available users from the left side to the paging/intercom group
Members
member list on the right.

Click on
to edit the paging/intercom group.
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
Click on

Click on "Paging/Intercom Group Settings" to edit Alert-Info Header. This header will be included in the
to delete the paging/intercom group.
SIP INVITE message sent to the callee in paging/intercom call.
Figure 136: Page/Intercom Group Settings

The UCM6510 has pre-configured paging/intercom feature code. By default, the Paging Prefix is *81
and the Intercom Prefix is *80. To edit page/intercom feature code, click on "Feature Codes" in the
"Paging/Intercom Group Settings" dialog. Or users could go to web GUI->PBX->Internal Options>Feature Codes directly.
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CALL QUEUE
The UCM6510 supports call queue by using static agents or dynamic agents. Call Queue system can
accept more calls than the available agents. Incoming calls will be hold until next representative is available
in the system. This sections describes the configuration of call queue under web GUI->PBX->Call
Features->Call Queue.
Configure Call Queue
Call queue settings can be accessed via web GUI->PBX->Call Features->Call Queue.

Click on "Create New Queue" to add call queue.
Figure 137: Call Queue

Click on
to edit the call queue. The call queue configuration parameters are listed in the table
below.
Table 75: Call Queue Configuration Parameters
Extension
Configure the call queue extension.
Name
Configure the call queue name to identify the call queue.
Select the strategy for the call queue.

Ring All
Ring all available Agents simultaneously until one answers.

Linear
Ring agents in the specified order.

Least Recent
Ring the agent who has been called the least recently.
Strategy

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.
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The default setting is "Ring All".
Select the Music On Hold class for the call queue.
Music On Hold
Note:
Music On Hold classes can be managed from web GUI-> PBX->Internal
Options->Music On Hold.
Configure whether the callers will be disconnected from the queue or not if
the queue has no agent anymore. The default setting is "Strict".

Yes
Callers will be disconnected from the queue if all agents are paused or
Leave When Empty
invalid.

No
Never disconnect the callers from the queue when the queue is empty.

Strict
Callers will be disconnected from the queue if all agents are paused,
invalid or unavailable.
Configure whether the callers can dial into a call queue if the queue has no
agent. The default setting is "No".

Yes
Callers can always dial into a call queue.
Dial in Empty Queue

No
Callers cannot dial into a queue if all agents are paused or invalid.

Strict
Callers cannot dial into a queue if the agents are paused, invalid or
unavailable.
Assign permission level to the user. The available permissions are
Permission
"Internal", "Local", "National" and "International" from the lowest level to the
highest level. The default setting is "Internal".
Dynamic Login Password
Ring Time Out
If enabled, the configured PIN number is required for dynamic agent to log
in. The default setting is disabled.
Configure the number of seconds an agent will ring before the call goes to
the next agent. The default setting is 15 seconds.
Configure the number of seconds before a new call can ring the queue after
Wrapup Time
the last call on the agent is completed. If set to 0, there will be no delay
between calls to the queue. The default setting is 15 seconds.
Retry Time
Configure the number of seconds to wait before ringing the next agent.
Configure the maximum number of calls to be queued at once. This number
does not include calls that have been connected with agents. It only
Max Queue Length
includes calls not connected yet. The default setting is 0, which means
unlimited. When the maximum value is reached, the caller will be treated
with busy tone followed by the next calling rule after attempting to enter the
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queue.
If enabled, the UCM6510 will report (to the agent) the duration of time of
Report Hold Time
the call before the caller is connected to the agent. The default setting is
"No".
If enabled, users will be disconnected after the configured number of
seconds. The default setting is "No".
Wait Time
Note:
It is recommended to configure "Wait Time" longer than the "Wrapup Time".
If this option is enabled, the calls using this extension or trunk will be
Auto Record
automatically recorded.
If enabled, the incoming call for the call queue will be routed to the
Enable Destination
destination configured in the next field if none of the agents answers the
call after ringing for a time of “Ring Timeout”.
Configure the global timeout (in seconds) of this call queue. It must be larger
than the value of ring timeout. The call will be transferred to fail over
Queue Timeout
destination directly if this time is exceeded.
The call would be routed to this destination if no one in this ring group
Failover Destination
answers the call. It can be set to Extension, Voicemail, Queues, Ring
Group, Voicemail Group, IVR, and External Number.
When present in an INVITE request, the alert-Info header field specifies and
Alert-Info
alternative ring tone to the UAS.
Enable Feature Codes
Enable feature codes option for call queue. For example, *83 is used for
“Agent Pause”
Select the available users to be the static agents in the call queue. Choose
Agents
from the available users on the left to the static agents list on the right. Click
on
to arrange the order.

Click on

Click on "Agent Login Settings" to configure Agent Login Extension Postfix and Agent Logout Extension
to delete the call queue.
Postfix. Once configured, users could log in the call queue as dynamic agent.
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Figure 138: Agent Login Settings
For example, if the call queue extension is 6510, Agent Login Extension Postfix is * and Agent Logout
Extension Postfix is **, users could dial 6510* to login to the call queue as dynamic agent and dial
6510** to logout from the call queue. Dynamic agent doesn't need to be listed as static agent and can
log in/log out at any time.

Call queue feature code "Agent Pause" and "Agent Unpause" can be configured under web GUI->PBX>Internal Options->Feature Codes. The default feature code is *83 for "Agent Pause" and *84 for
"Agent Unpause".

Queue recordings are shown on the Call Queue page. Click on
.wav format; click on
to download the recording file in
to delete the recording file. To delete multiple recording files by one click,
select several recording files to be deleted and click on “Delete Selected Recording Files” or click on
“Delete All Recording Files” to delete all recording files.
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EXTENSION GROUPS
The UCM6510 extension group feature allows users to assign extensions to different groups to better
manage the configurations on the PBX. For example, when configuring "Enable Filter on Source Caller ID",
users could select a group instead of each person's extension to assign. This feature simplifies the
configuration process and helps manage and categorize the extensions for business environment.
Configure Extension Groups
Extension group can be configured via web GUI->PBX->Call Features->Extension Groups.

Click on "Create New Extension Group" to create a new extension group.

Click on
to edit the extension group.
Select extensions from the list on the left side to the right side.
Figure 139: Edit Extension Group

Click on
to delete the extension group.
Use Extension Groups
Here is an example where the extension group can be used. Go to web GUI->PBX->Basic/Call Routes>Outbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and extension
groups will show up for users to select.
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Figure 140: Select Extension Group in Outbound Route
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PICKUP GROUPS
The UCM6510 supports pickup group feature which allows users to pick up incoming calls for other
extensions if they are in the same pickup group, by dialing "Pickup Extension" feature code (by default *8).
Configure Pickup Groups
Pickup groups can be configured via web GUI->PBX->Call Features->Pickup Groups.

Click on "Create New Pickup Group" to create a new pickup group.

Click on
to edit the pickup group.
Select extensions from the list on the left side to the right side.
Figure 141: Edit Pickup Group

Click on
to delete the pickup group.
Configure Pickup Feature Code
When picking up the call for the pickup group member, the user only needs to dial the pickup feature code.
It’s not necessary to add the extension number after the pickup feature code. The pickup feature code is
configurable under Web GUI->PBX->Internal Options->Feature Codes.
The default pickup feature code is *8.
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Figure 142: Edit Pickup Feature Code
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MUSIC ON HOLD
Music On Hold settings can be accessed via web GUI->PBX->Internal Options->Music On Hold. In this
page, users could configure music on hold class and upload music files. The "default" Music On Hold class
already has 5 audio files defined for users to use.
Figure 143: Music On Hold Default Class

Click on "Create New MOH Class" to add a new Music On Hold class.

Click on
to configure the MOH class sort method to be "Alpha" or "Random" for the sound files.

Click on
next to the selected Music On Hold class to delete this Music On Hold class.

Click on
to select music file from local PC and click on
to start uploading. The music file
uploaded has to be 16 bit, 8 KHz Mono in .wav format with size smaller than 5M.

Click on
next to the sound file to delete it from the selected Music On Hold Class.
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Note:
In case the users have deleted the system MOH files, there are two ways to recover.
1. 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, unzip the pack and upload the music files to the UCM6510.
2. Factory reset could also recover the MOH file on the UCM.
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FAX/T.38
The UCM6510 supports T.30/T.38 Fax and Fax Pass-through. It can also convert the received Fax to PDF
format and send it to the configured Email address. Fax/T.38 settings can be accessed via web GUI->PBX>Internal Options->FAX/T.38.
Configure Fax/T.38

Click on "Create New Fax Extension". In the popped up window, fill the extension, name and Email
address to send the received Fax to.

Click on "Fax Settings" to configure the Fax parameters.
Table 76: FAX/T.38 Settings
Enable Error Correction
Configure to enable Error Correction Mode (ECM) for the Fax. The default
Mode
setting is "Yes".
Configure the maximum transfer rate during the Fax rate negotiation. The
Maximum Transfer Rate
possible values are 2400, 4800, 7200, 9600, 12000 and 14400. The default
setting is 14400.
Configure the minimum transfer rate during the Fax rate negotiation. The
Minimum Transfer Rate
possible values are 2400, 4800, 7200, 9600, 12000 and 14000. The default
setting is 2400.
Configure the concurrent fax that can be sent by UCM6510.
Max Concurrent Sending Fax

Only mode supports single user fax sending

More mode supports multiple concurrent fax sending
By default, this option is set to “only”.
Fax Queue Length
Configure the maximum length of Fax Queue.
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 be not be received from Email.
Fill in the "Subject:" and "Message:" content, to be used in the Email when
Template Variables
sending the Fax to the users.
The template variables are:
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
${CALLERIDNUM} : Caller ID Number

${CALLERIDNAME} : Caller ID Name

${RECEIVEEXTEN} : The extension to receive the Fax

${FAXPAGES} : Number of pages in the Fax

${VM_DATE} : The date and time when the Fax is received

Click on
to edit the Fax extension.

Click on
to delete the Fax extension.
Sample Configuration to Receive Fax from PSTN Line
The following instructions describe how to use the UCM6510 to receive Fax from PSTN line on the Fax
machine connected to the UCM6510 FXS port.
1. Connect Fax machine to the UCM6510 FXS port.
2. Connect PSTN line to the UCM6510 FXO port.
3. Go to web GUI->PBX->Analog Trunks page.
4. Create or edit the analog trunk for Fax as below.
Fax Detection: Make sure "Fax Detection" option is set to "No".
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Figure 144: Configure Analog Trunk without Fax Detection
5. Go to UCM6510 web GUI->PBX->Basic/Call Routes->Extensions page.
6. Create or edit the extension for FXS port.

Analog Station: Select FXS port to be assigned to the extension. By default, it's set to "None".

Once selected, analog related settings for this extension will show up in "Analog Settings" section.
Figure 145: Configure Extension For Fax Machine
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Figure 146: Configure Extension for Fax Machine: Analog Settings
7. Go to web GUI->PBX->Basic/Call Routes->Inbound Routes page.
8. Create an inbound route to use the Fax analog trunk. Select the created extension for Fax machine in
step 4 as the default destination.
Figure 147: Configure Inbound Rule for Fax
Now the Fax configuration is done. When there is an incoming Fax calling to the PSTN number for the FXO
port, it will send the Fax to the Fax machine.
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Sample Configuration for Fax-To-Email
The following instructions describe a sample configuration on how to use Fax-to-Email feature on the
UCM6510.
1. Connect PSTN line to the UCM6510 FXO port.
2. Go to UCM6510 web GUI->Internal Options->Fax/T.38 page. Create a new Fax extension.
Figure 148: Create Fax Extension
3. Go to UCM6510 web GUI->Basic/Call Routes->Analog Trunks page. Create a new analog trunk with
"FAX Detection" set to "No".
4. Go to UCM6510 web GUI->Basic/Call Routes->Inbound Routes page. Create a new inbound route
and set the default destination to the Fax extension.
Figure 149: Inbound Route to Fax Extension
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5. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number
will be converted to PDF file and sent to the Email address Faxtest@ucm6510mycompany.com as
attachment.
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ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)
Starting from firmware 1.0.1.10, the UCM6510 supports Asterisk Manager Interface (AMI) with restricted
access. AMI allows a client program to connect to an Asterisk instance commands or read events over a
TCP/IP stream. It’s particularly useful when the system admin tries to track the state of a telephony client
inside Asterisk.
User could configure AMI parameters on UCM6510 web GUI->PBX->Internal Options->AMI. For details
on how to use AMI on UCM6510, please refer to the following AMI guide:
http://www.grandstream.com/sites/default/files/Resources/ucm6100_AMI_guide.pdf
--------------------------------------------------------------------------------------------------------------------- ----------------------Warning:
Please do not enable AMI on the UCM6510 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
UCM6510 system. Please be cautious when enabling AMI access on the UCM6510 and restrict the
permission granted to the AMI user. By using AMI on UCM6510 you agree you understand and
acknowledge the risks associated with this.
--------------------------------------------------------------------------------------------------------------------- -----------------------
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FOLLOW ME
Follow Me is a feature on the UCM6510 that allows users to direct calls to other phone numbers and have
them ring all at once or one after the other. Calls can be directed to users’ home phone, office phone, mobile
and etc. The calls will get to the user no matter where they are. Follow Me option can be found under web
GUI-> PBX-> Call Features-> Follow Me.
To configure follow me:
1. Click on "Create New Follow Me" and then select an extension to be configured with Follow Me.
Figure 150: Create Follow Me
2. Click on “Next” to continue editing Follow Me configuration.
Figure 151: Edit Follow Me
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3. Click on “Add Follow Me Number” to add local extensions or external numbers to be called after ringing
the extension selected in the first step.
4. Once created, it will be displayed on the follow me web page list. Click on
configuration. Click on
to edit the Follow Me
to delete the Follow Me.
The following table shows the Follow Me configuration parameters.
Table 77: Follow Me Settings
Enable
Configure to enable or disable Follow Me for this user.
If external number is added in the Follow Me, please make sure this option
Skip Trunk Auth
is enabled or the “Skip Trunk Auth” option of the extension is enabled,
otherwise the external Follow Me number cannot be reached.
Music On Hold Class
Configure the Music On Hold class that the caller would hear while tracking
the user.
By default, it is enabled and user will be asked to press 1 to accept the call
Confirm When Answering
or to press 2 to reject the call after answering a Follow Me call.
If it is disabled, the Follow Me call will be established once after the user
answers it.
Enable Destination
If enabled, the call will be routed to the default destination if no one in the
Follow Me answers the call.
Configure the destination if no one in the Follow Me answers the call,
available options are:
Default Destination
Follow Me Numbers

Extension

Voicemail

Queues

Ring Group

Voicemail Group

IVR

External Number
The added numbers are listed here. Click on
Click on
to delete the number. Click on
to arrange the order.
to add new numbers.
Add a new Follow Me number which could be a ‘Local Extension’ or
New Follow Me Number
‘External Number’. The selected dial plan should have permissions to dial
the defined external number.
Dialing Order
Select the order in which the Follow Me destinations will be dialed to reach
the user: ring all at once or ring one after the other.
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5. Click on “Follow Me Options” to enable or disable the options listed in the following table.
Table 78: Follow Me Options
Playback Incoming Status
If enabled, the PBX will playback the incoming status message before
Message
starting the Follow Me steps.
Record the Caller’s Name
If enabled, the PBX will record the caller’s name from the phone so it can
be announced to the callee in each step.
Playback Unreachable
If enabled, the PBX will playback the unreachable status message to the
Status Message
caller if the callee cannot be reached.
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ONE-KEY DIAL
The UCM6510 supports One-Key Dial that allows users to call a certain destination by pressing one digit 0
to 9 on the keypad. This creates a system-wide speed dial access for all the extensions on the UCM6510.
To enable One-Key Dial, on the UCM6510 web GUI, go to page PBX->Call Features->One-Key Dial.
Figure 152: Configure One-Key Dial
User should first decide a digit used for One-Key Dial and check the option “Enable Destination” for the
digit.
Then select a dial destination from “Default Destination”. The supported destinations include
extension, voicemail, conference room, voicemail group, IVR, ring group, call queue, page group, fax, DISA,
Dial by Name and external number.
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Figure 153: One-Key Dial Destinations
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DISA
In many situations the user will find the need to access his own IPPBX resources but he is not physically
close to any 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 first, whether it’s using his cell phone, pay phone, regular PSTN and
etc, and then call into a SIP trunk or PSTN trunk connected to UCM6510 as it is an internal extension.
The UCM6510 supports DISA to be used in IVR or inbound route. Before using it, create new DISA under
web GUI->Call Features->DISA.

Click on "Create New DISA" to add a new DISA.

Click on
to edit the DISA configuration.

Click on
to delete the DISA.
Figure 154: Create New DISA
Table 79: DISA Settings
Name
Configure DISA name to identify the DISA.
Configure the password (digit only) required for the user to enter before
using DISA to dial out.
Password
Note:
The password has to be at least 4 digits.
Permission
Configure the permission level for DISA. The available permissions are
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"Internal", "Local", "National" and "International" from the lowest level to the
highest level. The default setting is "Internal". If the user tries to dial
outbound calls after dialing into the DISA, the UCM6510 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 UCM6510 will wait before
Response Timeout
hanging up if the user dials an incomplete or invalid number. The default
setting is 10 seconds.
Digit Timeout
Configure the maximum amount of time permitted between digits when the
user is typing the extension. The default setting is 5 seconds.
If enabled, during an active call, users can enter the UCM6510 hangup
Allow Hangup
feature code (*0 by default) to disconnect the call or hang up directly. A new
dial tone will be heard shortly for the user to make a new call. The default
setting is "No".
Once successfully created, users can configure the inbound route destination as "DISA" or IVR key event
as "DISA". When dialing into DISA, users will be prompted with password first. After entering the correct
password, a second dial tone will be heard for the users to dial out.
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CALLBACK FEATURE
Callback is mainly designed for users who often use their mobile phones to make long distance or
international calls which may have high service charges. The callback feature provides an economic
solution for reduce the cost from this.
The callback feature works as follows:
1. Configure a new callback on the UCM6510.
2. On the UCM6510, 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 UCM6510 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 UCM6510 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 UCM6510 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 UCM6510, go to web GUI->PBX->Call Features->Callback page and click
on
. Configuration parameters are listed in the following table.
Table 80: Callback Configuration Parameters
Name
Configure a name to identify the Callback.
Configure the pattern of the callers allowed to use this callback. The caller
who places the inbound call needs to have the callerID match this pattern
CallerID Pattern
so that the caller can get callback after hanging up the call.
Note:
If leaving as blank, all numbers are allowed to use this callback.
Configure the prepend digits to be added at before dialing the outside
Outbound Prepend
number. 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.
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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 UCM6510
trunk.
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BLF AND EVENT LIST
BLF
The UCM6510 supports BLF monitoring for extensions, ring group, call queue, conference room and
parking lot. For example, on the user's phone, configure the parking lot number 701 as the BLF monitored
number. When there is a parked call on 701, the LED for this BLF key will light up in red, meaning a call is
parked against this parking lot. Pressing this BLF key can pick up the call from this parking lot.
-------------------------------------------------------------------------------------------------------------------------------------------Note:

On the Grandstream GXP phones, the MPK supports "Call Park" mode, which is normally used to park
the call by configuring the MPK number as call park feature code (e.g., 700). Users could also use "Call
Park" mode to monitor and pick up the call on this parking lot by configuring the MPK number as parking
lot number (e.g., 701).
----------------------------------------------------------------------------------------------------------- ---------------------------------
Event List
Besides BLF, users can also configure the phones to monitor event list. By using event list, local extensions
on the same UCM6510 or remote extensions on the VOIP trunk can be monitored. The event list settings
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 155: Create New Event List
Table 81: Event List Settings
Configure the name of this event list (for example, office_event_list). Please
URI
note the URI name cannot be the same as the extension name on the
UCM6510. The valid characters are letters, digits, _ and -.
Local Extensions
Select the available extensions listed on the local UCM6510 to be
monitored in the event list.
If LDAP sync is enabled between the UCM6510 and the peer UCM6510,
Remote Extensions
the remote extensions will be listed under "Available Extensions". If not,
manually enter the remote extensions under "Special Extensions" field.
Manually enter the remote extensions in the peer/register trunk to be
Special Extensions
monitored in the event list.
Valid format: 5000,5001,9000
Remote extension monitoring works on the UCM6510 via event list BLF, among Peer SIP trunks or Register
SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the UCM6510 first
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before using remote BLF feature. Please note the SIP end points need support event list BLF in order to
monitor remote extensions.
When an event list is created on the UCM6510 and remote extensions are added to the list, the UCM6510
will send out SIP SUBSCIRBE to the remote UCM6510 to obtain the remote extension status. When the
SIP end points registers and subscribes to the local UCM6510 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.
--------------------------------------------------------------------------------------------------------- ----------------------------------Note:

To configure LDAP sync, please go to UCM6510 web GUI->PBX->Basic/Call Routes->VoIP Trunk.
You will see "Sync LDAP Enable" option. Once enabled, please configure password information for the
remote peer UCM6510 to connect to the local UCM6510. Additional information such as port number,
LDAP outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM6510 and remote
UCM6510 need enable LDAP sync option with the same password for successful connection and
synchronization.

Currently LDAP sync feature only works between two UCM6510s.

(Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is nonUCM6510 PBX. However, it might not work the other way around depending on whether the nonUCM6510 PBX supports event list BLF or remote monitoring feature.
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DIAL BY NAME
Dial By Name is a feature on the PBX that allows caller to search a person by first or last name via his/her
phone's keypad. The administrator can define the Dial By Name directory including the desired extensions
in the directory and the searching type by "first name" or "last name". After dialing in, the PBX IVR/Auto
Attendant will guide the caller to spell the digits to find the person in the Dial By Name directory. This feature
allows customers/clients to use the guided automatic system to get in touch with the enterprise employees
without having to know the extension number, which brings convenience and improves business image for
the enterprise.
Dial By Name Configuration
The administrators can create the dial by name group under web GUI->PBX->Call Features->Dial By
Name.
Figure 156: Create Dial By Name Group
1. Group Name
Enter the Group Name. This is to identify the Dial By Name group. The Dial By Name group can be
used as the destination for inbound route and key pressing event for IVR. The group name defined here
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will show up in the destination list when configuring IVR and inbound route.
Figure 157: Dial By Name Group In IVR Key Pressing Events
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Figure 158: Dial By Name Group In Inbound Route
2. Extension
Configure the direct dial extension for the Dial By Name group.
3. Available Extensions/Selected Extensions
Select available extensions from the left side to the right side as the directory for the Dial By Name
group. Only the selected extensions here can be reached by the Dial By Name IVR when dialing into
this group. The extensions here must have valid first name and last name configured under web GUI>PBX->Basic/Call Routes->Extensions in order to be searchable in Dial By Name directory through
IVR. By specifying the extensions here, the administrators can make sure unscreened calls will not
reach the company employee if he/she doesn't want to receive them directly.
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Figure 159: Configure Extension First Name and Last Name
4. Query Type
Specify the query type. This defines how the caller will need to enter to search the directory.
By First Name: enter the first 3 digits of the first name to search the directory.
By Last Name: enter the first 3 digits of the last name to search the directory.
By Full Name: enter the first 3 digits of the first name or last name to search the directory.
5. Select Type
Specify the select type on the searching result. The IVR will confirm the name/number for the party the
caller would like to reach before dialing out
By Order: After the caller enters the digits, the IVR will announce the first matching party's name and
number. The caller can confirm and dial out if it's the destination party, or press * to listen to the next
matching result if it's not the desired party to call.
By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can
press number 1 to 8 to select and call, or press 9 for results in next page.
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ACTIVE CALLS AND MONITOR
The active calls on the UCM6510 are displayed in web UI->Status->Active Calls page. Users can monitor
call status, hang up active call(s) as well as barge in active call(s) in real time manner.
Active Calls Status
To view the status of active calls, navigate to web GUI->Status->Active Calls. The following figure shows
extension 1000 is calling 1001. 1001 is ringing.
Figure 160: Status->PBX Status->Active Calls - Ringing
The following figure shows the call between 1000 and 1001 is established.
Figure 161: Status->PBX Status->Active Calls – Call Established
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On Active Calls page, click on
to refresh the status of active calls.
The green color of the active call means the connection of call time is less than half an hour. It means this
call is normal.
Figure 162: 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 163: call connection between half an hour and one hour
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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.
Figure 164: 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 shown on the Active Calls page.
Call Monitor
During an active call, click on icon
and the Monitor dialog will pop up.
Figure 165: Configure to Monitor an Active Call
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In the “Monitor” dialog, configure the following to monitor an active call:
1. Enter an available extension for “Monitor’s Extension” which will be used to monitor the active call.
2. “Monitored Extension” must be one of the parties in the active call to be monitored.
3. Select spy mode. There are three options in “Spy Mode”.

Listen
In “Listen” mode, the extension monitoring the call can hear both parties in the active call but the
audio of the user on this extension will not be heard by either party in the monitored active call.

Whisper
In “Whisper” mode, the extension monitoring the call can hear both parties in the active call. The
user on this extension can only talk to the selected monitored extension and he/she will not be
heard by the other party in the active call. This can be usually used to supervise calls.

Barge
In “Barge” mode, the extension monitoring the call can talk to both parties in the active call. The
call will be established similar to three-way conference.
4. Enable or disable “Require Confirmation” option. If enabled, the confirmation of the invited monitor’s
extension is required before the active call can be monitored. This option can be used to avoid adding
participant who has auto-answer configured or call forwarded to voicemail.
5. Click on “Add”. An INVITE will be sent to the monitor’s extension. The monitor can answer the call and
start monitoring. If “Require Confirmation” is enabled, the user will be asked to confirm to monitor the
call.
Another way to monitor active calls is to dial the corresponding feature codes from an extension. Please
refer to [Table 82: UCM6510 Feature Codes] and [Enable Spy] section for instructions.
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CALL FEATURES
The UCM6510 supports call recording, transfer, call forward, call park and other call features via feature
code. Feature Codes settings can be found at web GUI->PBX->Internal Options->Feature Codes. This
section lists all the feature codes in the UCM6510 and describes how to use the call features.
Feature Codes
Table 82: UCM6510 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
Attended Transfer
the attended transfer.

Options:
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.

Default code: *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.
Call Park

Default code: #72.

Enter the code during active call to park the call.

Options:
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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 UCM6510 will mix the streams natively on the fly as
the call is in progress.
Audio Mix Record

Options:
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.
DND/Call Forward
Do Not Disturb (DND) Activate

Default code: *77.
Do Not Disturb (DND) Deactivate

Default code: *78.

Default Code: *90.

Enter the code and follow the voice prompt. Or enter the code
Call Forward Busy Activate
followed by the extension to forward the call.
Call Forward Busy Deactivate
Call Forward No Answer
Activate
Call Forward No Answer
Deactivate
Call Forward Unconditional
Activate
Call Forward Unconditional
Deactivate

Default Code: *91.

Default Code: *92.

Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.

Default Code: *93.

Default Code: *72.

Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.

Default Code: *73.

Default Setting: 1000.

Configure the maximum interval (in milliseconds) between the
Feature Misc
Feature Code Digits Timeout
digits input to activate the feature code.
Call Park

Default Extension: 700.

During an active call, initiate blind transfer and then enter this
code to park the call.
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Parked Lots

Default Extension: 701-720.

These are the extensions where the calls will be parked, i.e.,
parking lots that the parked calls can be retrieved.
If checked, the parklot number will be used as extension, you can
Use parklot as extension
transfer to the parklot number for parking your call. If the parking lots
overlap existing extensions, there will be conflict.
Parking Timeout (s)

Default setting: 300.

This is the timeout allowed for a call to be parked. After the
timeout, if the call is not picked up, the extension who parks the
call will be called back.
Feature Codes
Voicemail Access Code

Default Code: *98.

Enter *98 and follow the voice prompt. Or dial *98 followed by the
extension and # to access the entered extension's voicemail box.
My Voicemail
Agent Pause
Agent Unpause
Paging Prefix

Default Code: *97.

Press *97 to access the voicemail box.

Default Code: *83.

Pause the agent in all call queues.

Default Code: *84.

Unpause the agent in all call queues.

Default Code: *81.

To page an extension, enter the code followed by the extension
number.
Intercom Prefix

Default Code: *80.

To intercom an extension, enter the code followed by the
extension number.
Blacklist Add

Default Code: *40.

To add a number to blacklist for inbound route, dial *40 and follow
the voice prompt to enter the number.
Blacklist Remove

Default Code: *41.

To remove a number from current blacklist for inbound route, dial
*41 and follow the voice prompt to remove the number.
Call Pickup on Ringing

Default Code: **.

To pick up a call for any extension xxxx, enter the code followed
by the extension number xxxx.
Pickup Extension

Default Code: *8.

This code is for the pickup group which can be assigned for each
extension on the extension configuration page.

If there is an incoming call to an extension, the other extensions
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within the same pickup group can dial *8 directly to pick up the
call.

Default Code: *

This code is for the user to directly dial or transfer to an
extension's voicemail.
Direct Dial Voicemail Prefix

For example, directly dial *5000 will have to call go into the
extension 5000's voicemail. If the user would like to transfer the
call to the extension 5000's voicemail, enter *5000 as the transfer
target number.
Call Completion Request

Default Code: *11

This code is for the user who wants to use Call Completion to
complete a call.
Call Completion Cancel

Default Code: *12

This code is for the user who wants to cancel Call Completion
request.
Enable Spy

Check this box to enable spy feature codes.

This is the feature code to listen in on a call to monitor
performance. Monitor’s line will be muted, and neither party will
Listen Spy
hear from the monitor’s extension. The default setting is *54.

This is the feature code to speak to one side of the call (for
example, whisper to employees to help them handle a call). Only
Whisper Spy
one side will be able to hear from the monitor’s extension. The
default setting is *55.
Barge Spy
Enable Inbound Multiple Mode
Inbound Default Mode
Inbound Mode 1

This is the feature code to join in on the call to assist both parties.
The default setting is *56.
If enabled, user can switch between different inbound route
modes with feature code. By default, this option is disabled.
This feature code is used to switch inbound route mode to default
mode. The default setting is *61.
This feature code is used to switch inbound route mode to mode
1. The default setting is *62.
The UCM6510 also allows user to one click enable / disable specific feature code. As shown below:
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Figure 166: Enable/Disable Feature codes
Call Recording
The UCM6510 allows users to record audio during the call. If "Auto Record" is turned on for extension or
trunk, the call will be automatically recorded when there is established call with the extension or trunk.
Otherwise, please follow the instructions below to manually record the call.
1. Make sure the feature code for "Audio Mix Record" is configured and enabled.
2. After establishing the call, enter the "Audio Mix Record" feature code (by default *3) followed by # or
SEND to start recording.
3. To stop the recording, enter the "Audio Mix Record" feature code (by default *3) followed by # or SEND
again. Or the recording will be stopped once the call hangs up.
4. The recording file can be retrieved under web GUI->Status->CDR. Click on
or click on
to play the recording
to download the recording file.
Figure 167: Download Recording File from CDR Page
The above recorded call's recording files are also listed under the UCM6510 web GUI->CDR->Recording
Files.
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Call Park
The UCM6510 provides call park and call pickup features via feature code.
Park a Call
There are two feature codes that can be used to park the call.

Feature Maps->Call Park (Default code #72)
During an active call, press #72 and the call will be parked. Parking lot number (default range 701 to
720) will be announced after parking the call.

Feature Misc->Call Park (Default code 700)
During an active call, initiate blind transfer (default code #1) and then dial 700 to park the call. Parking
lot number (default range 701 to 720) will be announced after parking the call.
Retrieve The Parked Call
To retrieve the parked call, simply dial the parking lot number and the call will be established. If a parked
call is not retrieved after the timeout, the original extension who parks the call will be called back.
Enable Spy
If “Enable Spy” option is enabled, feature codes for Listen Spy, Whisper Spy and Barge Spy are available
for users to dial from any extension to perform the corresponding actions.
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.
--------------------------------------------------------------------------------------------------------------------- ----------------------Warning:
“Enable Spy” allows any user to listen to any call by dialing feature codes. This may result in the leakage
of user privacy. Please be aware of the associated potential security risk when enabling this feature.
--------------------------------------------------------------------------------------------------------------------- -----------------------
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INTERNAL OPTIONS
This section describes internal options that haven't been mentioned in previous sections yet. The settings
in this section can be applied globally to the UCM6510, including general configurations, jitter buffer, RTP
settings, hardware config and STUN monitor. The options can be accessed via web GUI->PBX->Internal
Options.
Internal Options/General
Table 83: Internal Options/General
General Preferences
Configure the global CallerID used for all outbound calls when no other
Global OutBound CID
CallerID 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
Global OutBound CID Name
configured, all outbound calls will have the CallerID Name set to this name.
If not, the extension's CallerID Name will be used.
Specify the operator extension, which will be dialed when users press 0 to
Operator Extension
exit voicemail application. The operator extension can also be used in IVR
option.
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
Record Prompt
stopped. For example, before recording, the UCM6510 will play voice
prompt "The call will be recorded". The default setting is "No".
Extension Preferences
If enabled, strong password will be enforced for the password created on
the UCM6510. The default setting is enabled.
Enforce Strong Passwords
Strong Password Rules:
1. Password for voicemail, voicemail group, outbound route, DISA, call
queue and conference requires non-repetitive and non-sequential
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digits, with a minimum length of 4 digits. Repetitive digits pattern (such
as 0000, 1111, 1234, 2345, and etc), or common digits’ pattern (such
as 111222, 321321 and etc) are not allowed to be configured as
password.
2. Password for extension registration, web GUI admin login, LDAP and
LDAP sync requires alphanumeric characters containing at least two
categories of the following, with a minimum length of 4 characters.

Numeric digits

Lowercase alphabet characters

Uppercase alphabet characters

Special characters
If enabled, random password will be generated when the extension is
Enable Random Password
created. The default setting is "Yes". It is recommended to enable it for
security purpose.
Enable Auto Email To User
If enabled, UCM6510 will send Email notification to user automatically after
editing extension settings or adding a new extension.
If set to "Yes", users could disable the extension range preconfigured/configured on the UCM6510. The default setting is "No".
The default extension range assignment is shown in “Extension Ranges”
Disable Extension Range
below.
Note:
It is recommended to keep the system assignment to avoid inappropriate
usage and unnecessary issues.
The default extension range assignment is:

User Extensions: 1000-6299
User Extensions is referring to the extensions created under web UI>PBX->Basic/Call Routes->Extensions page.

Pick Extensions: 4000-4999
This refers to the extensions that can be manually picked from end
Extension Ranges
device when being provisioned by the UCM6510. There are two related
options in zero config page->Auto Provision Settings, "Pick Extension
Segment" and "Enable Pick Extension". If "Enable Pick Extension"
under zero config settings is selected, the extension list defined in "Pick
Extension Segment" will be sent out to the device after receiving the
device's request. This "Pick Extension Segment" should be a subset of
the "Pick Extensions" range here. This feature is for the GXP series
phones that support selecting extension to be provisioned via phone's
LCD.
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
Auto Provision Extensions: 5000-6299
This sets the range for "Zero Config Extension Segment" which is the
extensions can be assigned on the UCM6510 to provision the end
device.

Conference Extensions: 6300-6399

Ring Group Extensions: 6400-6499

Queue Extensions: 6500-6599

Voicemail Group Extensions: 6600-6699

IVR Extensions: 7000-7100

Dial By Name Extensions: 7101-7199

Fax Extensions: 7200-8200
Internal Options/Jitter Buffer
Table 84: Internal Options/Jitter Buffer
SIP Jitter Buffer
Enable Jitter Buffer
Select to enable jitter buffer on the sending side of the SIP channel. The
default setting is "No".
Configure the time (in ms) to buffer. This is the jitter buffer size used in
Jitter Buffer Size
"Fixed" 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".

Fixed
The size is always equal to the value of "Max Jitter Buffer".
Implementation

Adaptive
The size is adjusted automatically and the maximum value equals to
the value of "Max Jitter Buffer".
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Internal Options/RTP Settings
Table 85: Internal Options/RTP Settings
RTP Start
Configure the RTP port starting number. The default setting is 10000.
RTP End
Configure the RTP port ending address. The default setting is 20000.
Configure to enable or disable strict RTP protection. If enabled, RTP
Strict RTP
packets that do not come from the source of the RTP stream will be
dropped. The default setting is "Disable".
RTP Checksums
Configure to enable or disable RTP Checksums on RTP traffic. The default
setting is "Disable".
Configure whether to support ICE, ICE is the integrated use of STUN and
TURN structure to provide reliable VoIP or video calls and media
ICE Support
transmission, via a SIP request/ response model or multiple candidate
endpoints exchanging IP addresses and ports, such as private addresses
and TURN server address. It is enabled by default.
Configure STUN server address, STUN protocol is a Client / Server – is
also a Request / Response protocol, where it is used to check the
connectivity between the two terminals, such as maintaining a NAT binding
entries keep alive agreement.
STUN Server
The default STUN Server is stun.ipvideotalk.com
Valid format:
[(hostname | IP-address) [':' port]
The default port number is 3478 if not specified.
Internal Options/Payload
The UCM6510 payload type for audio codecs and video codes can be configured here.
Table 86: Internal Options/Payload
AAL2-G.726
Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword
packing). The default setting is 112.
DTMF
Configured payload type for DTMF. The default setting is 101.
G.721 Compatible
Configure to enable/disable G.721 compatible. The default setting is Yes.
G.726
Configure the payload type for G.726 if "G.721 Compatible" is disabled.
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The default setting is 111.
iLBC
Configure the payload type for iLBC. The default setting is 97.
H.264
Configure the payload type for H.264. The default setting is 99.
H.263P
Configure the payload type for H.263+. The default setting is 100 103.
VP8
Configure the payload type for VP8. The default settings is 108.
Internal Options/PIN Groups
The UCM6510 supports pin group. Once pin group 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 under “ PBX->Internal Options->PIN Groups ”.
Table 87: PIN Group
Name
Specify the name of the group
Record In CDR
Specify whether to enable/disable record in CDR
PIN Number
Specify the code that will asked once dialing via a trunk
PIN Name
Specify the name of the PIN
Once user click on
the following figure pop’s up for configuring the new Pin.
Figure 168: Create New PIN Group
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Once PIN Groups and members Created it should look like:
Figure 169: PIN members
Please note, if pin group is enabled on outbound route level, password, privilege level and enable filter on
source caller ID will be disabled.
Figure 170: Outbound PIN
If pin group CDR is enabled, the call with pin group information will be displayed as part of CDR under
Account Code field.
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Figure 171: CDR Record
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IAX SETTINGS
The UCM6510 IAX global settings can be accessed via web GUI->PBX->IAX Settings.
IAX Settings/General
Table 88: IAX Settings/General
Bind Port
Bind Address
IAX1 Compatibility
Configure the port number that the IAX2 will be allowed to listen to. The
default setting is 4569.
Configure the address that the IAX2 will be forced to bind to. The default
setting is 0.0.0.0, which means all addresses.
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 features. The default setting
is "No".
Delay Reject
ADSI
If enabled, the IAX2 will delay the rejection of calls to avoid DOS. The
default setting is "No".
Select to enable ADSI phone compatibility. The default setting is "No".
Specify which Music On Hold class this channel would like to listen to when
Music On Hold Interpret
being put on hold. This music class is only effective if this channel has no
music class configured and the bridged channel putting the call on hold has
no "Music On Hold Suggest" setting.
Music On Hold Suggest
Bandwidth
Specify which Music On Hold class to suggest to the bridged channel when
putting the call on hold.
Configure the bandwidth for IAX settings. The default setting is "Low".
IAX Settings/Registration
Table 89: IAX Settings/Registration
IAX Registration Options
Min Reg Expire
Max Reg Expire
IAX Thread Count
IAX Max Thread Count
Configure the minimum period (in seconds) of registration. The default
setting is 60.
Configure the maximum period (in seconds) of registration. The default
setting is 3600.
Configure the number of IAX helper threads. The default setting is 10.
Configure the maximum number of IAX threads allowed. The default setting
is 100.
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If set to "yes", the connection will be terminated if ACK for the NEW
Auto Kill
message is not received within 2000ms. Users could also specify number
(in milliseconds) in addition to "yes" and "no". The default setting is "yes".
Authentication Debugging
If enabled, authentication traffic in debugging will not show. The default
setting is "No".
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
Trunk Time Stamps
Configure the frequency of trunk frames (in milliseconds). The default
setting is 20.
If enabled, time stamps will be attached to trunk frames. The default setting
is "No".
IAX Settings/Static Defense
Table 90: IAX Settings/Static Defense
Enter a single IP address or a range of IP addresses for which call token
Call Token Optional
validation is not required. For example:
11.11.11.11
11.11.11.11/22.22.22.22.
Max Call Numbers
Max Unvalidated Call
Numbers
Call Number Limits
Configure the maximum number of calls allowed for a single IP address.
Configure the maximum number of unvalidated calls for all IP addresses.
Configure to limit the number of calls for a give IP address of IP range.
Enter the IP address or a range of IP addresses to be considered for call
IP or IP Range
number limits. For example:
11.11.11.11
11.11.11.11/22.22.22.22.
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SIP SETTINGS
The UCM6510 SIP global settings can be accessed via web GUI->PBX->SIP Settings.
SIP Settings/General
Table 91: SIP Settings/General
Realm For Digest
Authentication
Bind UDP Port
Bind IP Address
Configure the host name or domain name for the UCM6510. Realms MUST
be globally unique according to RFC3261. The default setting is
grandstream.
Configure the UDP port used for SIP. The default setting is 5060.
Configure the IP address to bind to. The default setting is 0.0.0.0, which
means binding to all addresses.
If enabled, the UCM6510 allows unauthorized INVITE coming into the PBX
and the call can be made. The default setting is "No".
Allow Guest Calls
Warning:
Please be aware of the potential security risk when enabling "Allow Guest
Calls" as this will allow any user with the UCM6510 address to dial into the
UCM6510.
Allow Transfer
If set to "No", all transfers initiated by the endpoint in the UCM6510 will be
disabled (unless enabled in peers or users). The default setting is "Yes".
When sending MWI NOTIFY requests, this value will be used in the "From:"
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.
SIP Settings/Misc
Table 92: 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 UCM6510 gives
Register Attempts
up. The default setting is 0, which means the UCM6510 will keep trying until
the server side accepts the registration request.
Video
Max Bit Rate (kb/s)
Configure the maximum bit rate (in kb/s) for video calls. The default setting
is 384.
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Support SIP Video
Select to enable video support in SIP calls. The default setting is "Yes".
If enabled, when rejecting an incoming INVITE or REGISTER request, the
UCM6510 will always reject with "401 Unauthorized" instead of notifying the
Reject Non-Matching INVITE
requester whether there is a matching user or peer for the request. This
reduces the ability of an attacker to scan for valid SIP usernames. The
default setting is "No".
SIP Settings/Session Timer
Table 93: SIP Settings/Session Timer
Select the session timer mode. The default setting is "Accept".
The options are:

Session Timers
Originate
Always request and run session timer.

Accept
Run session timer only when requested by other UA.

Refuse
Do not run session timer.
Session Expire
Min SE
Session Refresher
Configure the maximum session refresh interval (in seconds). The default
setting is 1800.
Configure the minimum session refresh interval (in seconds). The default
setting is 90.
Select the session refresher to be UAC or UAS. The default setting is UAC.
SIP Settings/TCP and TLS
----------------------------------------------------------------------------------------------------------------------------- --------------Note:
The configuration in this section requires system reboot to take effect.
--------------------------------------------------------------------------------------------------------------------- ----------------------Table 94: SIP Settings/TCP and TLS
TCP Enable
Configure to allow incoming TCP connections with the UCM6510. The
default setting is "No".
Configure the IP address for TCP server to bind to. 0.0.0.0 means binding
TCP Bind Address
to all interfaces. The port number is optional. If not specified, 5060 will be
used.
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TLS Enable
Configure to allow incoming TLS connections with the UCM6510. The
default 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.
TLS Bind Address
Note:
The IP address must match the common name (hostname) in the certificate.
Please do not bind a TLS socket to multiple IP addresses. For details on
how to construct a certificate for SIP, please refer to the following document:
http://tools.ietf.org/html/draft-ietf-sip-domain-certs
TLS Client Protocol
TLS Do Not Verify
Select the TLS protocol for outbound client connections. The default setting
is TLSv1.
If enabled, the TLS server's certificate won't be verified when acting as a
client. The default setting is "Yes".
This is the CA certificate if the TLS server being connected to requires selfsigned certificate, including server's public key. This file will be renamed as
TLS Self-Signed CA
"TLS.ca" automatically.
Note:
The size of the uploaded ca file must be under 2MB.
This is the Certificate file (*.pem format only) used for TLS connections. It
contains private key for client and signed certificate for the server. This file
TLS Cert
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
TLS CA Cert
servers.
Note:
The size of the uploaded CA certificate file must be under 2MB.
TLS CA List
Display a list of files under the CA Cert directory.
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SIP Settings/NAT
Table 95: SIP Settings/NAT
Configure a static IP address and port (optional) used in outbound SIP
External Host
messages if the UCM6510 is behind NAT. If it is a host name, it will only be
looked up once.
Use IP address in SDP
External TCP Port
External TLS Port
If enabled, the SDP connection will use the IP address resolved from the
external host.
Configure the externally mapped TCP port when the UCM6510 is behind a
static NAT or PAT.
Configures the externally mapped TLS port when UCM6510 is behind a
static NAT or PAT.
Specify a list of network addresses that are considered inside of the NAT
network. Multiple entries are allowed. If not configured, the external IP
Local Network Address
address will not be set correctly.
A sample configuration could be as follows:
192.168.0.0/16
SIP Settings/TOS
Table 96: SIP Settings/ToS
ToS For SIP
ToS For RTP Audio
ToS For RTP Video
Configure the Type of Service for SIP packets. The default setting is None.
Configure the Type of Service for RTP audio packets. The default setting is
None.
Configure the Type of Service for RTP video packets. The default setting is
None.
Default Incoming/Outgoing
Configure the default duration (in seconds) of incoming/outgoing
Registration Time
registration. The default setting is 120.
Max
Registration/Subscription
Time
Min
Registration/Subscription
Time
Enable Relaxed DTMF
Configure the maximum duration (in seconds) of incoming registration and
subscription allowed by the UCM6510. The default setting is 3600.
Configure the minimum duration (in seconds) of incoming registration and
subscription allowed by the UCM6510. The default setting is 60.
Select to enable relaxed DTMF handling. The default setting is "No".
Select DTMF mode to send DTMF. The default setting is RFC2833. If "Info"
DTMF Mode
is selected, SIP INFO message will be used. If "Inband" is selected, 64-kbit
codec PCMU and PCMA are required. When "Auto" is selected, "RFC2833"
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will be used if offered, otherwise "Inband" will be used. The default setting
is "RFC2833".
During an active call, if there is no RTP activity within the timeout (in
seconds), the call will be terminated. The default setting is no timeout.
RTP Timeout
Note:
This setting doesn't apply to calls on hold.
When the call is on hold, if there is no RTP activity within the timeout (in
RTP Hold Timeout
seconds), the call will be terminated. This value of RTP Hold Timeout should
be larger than RTP Timeout. The default setting is no timeout.
Trust Remote Party ID
Send Remote Party ID
Configure whether the Remote-Party-ID should be trusted. The default
setting is "No".
Configure whether the Remote-Party-ID should be sent or not. The default
setting is "No".
Configure whether the UCM6510 should generate inband ringing or not.
The default setting is "Never".

Yes: The UCM6510 will send 180 Ringing followed by 183 Session
Progress and in-band audio.
Generate In-Band Ringing

No: The UCM6510 will send 180 Ringing if 183 Session Progress has
not been sent yet. If audio path is established already with 183 then
send in-band ringing.

Never: Whenever ringing occurs, the UCM6510 will send 180 Ringing
as long as 200OK has not been set yet. Inband ringing will not be
generated even the end point device is not working properly.
Server User Agent
Configure the user agent string for the UCM6510.
Send Compact SIP Headers
If enabled, compact SIP headers will be sent. The default setting is "No".
100rel
Configure the 100rel setting on UCM6510. The default setting is “Yes”.
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VALUE-ADDED FEATURES
Fax Sending
The UCM6510 supports sending Fax via web UI access. This feature can be found on web UI->PBX>Value-added Features->Fax Sending page. In order to send fax, pre-setup for analog trunk and
outbound route is required.
After making sure analog trunk or VoIP Trunk is setup properly and UCM6510 can reach out to PSTN
numbers via the trunk, on Fax Sending page, enter the fax number and upload the file to be faxed. Then
click on “Send” to start. The progress of sending fax will be displayed in web UI. Users can also view the
sending history is in the same web page.
Figure 172: Fax Sending in Web UI
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Announcements Center
UCM6510 series supports Announcements Center features start from firmware 1.0.2.7. Announcements
Center allows user to previously record and store voice message into UCM6510 with a specified code, and
user can also create groups with specified extensions. When code and group number is dialed together,
the specified voice message is sent to all group members and only extensions in the group will hear the
voice message.
Note: The Announcements Center is used in the combination with code + group number.
Figure 173: Announcements Center
Announcements Center Setting
Table 97: Announcements Center Setting
Name
Configure a name for the newly created Announcements Center, the name
can be anything.
Enter a code number for the customer prompt. This code is used in
combination with group number. For example, if the code is 55, and group
Code
number is 666. User dial 55666 will send prompt 55 to all members in
group 666.
Note: the final number must not conflict with any number, like extensions
or conference number.
This option is for setting a custom prompt as an announcement to notify
Custom Prompt
group members. The file can be uploaded from page ‘Custom Prompt’.
Click ‘Prompt’ to add additional record.
Ring Timeout
Configure the ring timeout for the group members. The default value is 30
seconds.
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Group Setting
Table 98: Group Setting
Configure a name for the newly created group, the name can be anything.
Name
Configure the group number. The group number is used in combination
with code. For example, if group number is 666, and code is 55. User dials
Number
55666 will send prompt 55 to all members in group 666.
Note: The group number must not conflict with any other numbers, such
as extension or conference number.
Announcements Center feature can be found under web UI-> PBX-> Value-added Features->
Announcements Center. The following example demonstrates the usage of this feature.
1. Click
to create new group.
2. Give a name for the newly created group, and the Number is used later with code to send voice
message.
3. Select the extensions that want to be included in the group.
Figure 174: Announcements Center Group Configuration
In this example, Group Test has number 666, and extension 1000, 1001 and 1002 is in the current group.
4. Click
to create a new Announcement Center.
5. Give a name for the newly created Announcements Center, and the Code is used to specifiy the
message that will be sent to the group.
6. Select the message that will be used by the code from the Custom Prompt drop down menu. To
create a new Prompt, please click Prompt and follow the instruction.
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Figure 175: Announcements Center Code Configuration
Code and Group number are used together to direct specified message to the target group. All extensions
in the group will received the message. For example, in this example, we can send code 55 to group 666.
To use, we can simply pick up any extension registered at UCM6510, dial number 55666 which is the
code with the group number. Extension 1000, 1001 and 1002 will receive this voice message.
Figure 176: Announcements Center example
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PMS (Property Management System)
UCM6510 support Hotel Property Management System PMS, including checkIn/checkout services, wakeup calls,
room status, Do Not Disturb which provide an ease of management for hotel application, this feature can be found
on web UI->PBX->PMS.
Note: The PMS integration on UCM was made with HMobile for now, more PMS software can be supported in
future releases.
In order to use all PMS features Please activate the feature code associated under “PBX->Internal Options>Feature Codes”

Enable PMS

Update PMS Room Status

PMS Wake Up Service Activate

PMS Wake Up Service Deactivate
Basic Settings
On the UCM WebGUI under “PBX->PMS->Basic Settings” set the connection information for the HMobile
platform.
Table 99: PMS Basic Settings
Field
Description
Prompt used when answering the wakeup calls it can
Wake Up Prompt
be customized from “PBX>Internal Options>Custom
Propmpt
PMS URL
Enter the PMS system URL
UCM Port
Enter the Port used by the PMS system
Username
Enter the Username to connect to the PMS system
Password
Enter the password to connect to the PMS system
Room Status
User can create Rooms by clicking on
, the following Figure will be displayed then.
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Figure 177: Create New Room
Click “Save” to create the new room, the fields above can be configured from the HMobile platform, once
set the following screen will be shown:
Figure 178: Room Status
User can Create a Batch of rooms as well by clicking on
, the following window will
pop up:
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Figure 179: Add batch rooms
Wake Up Service
In order to create a New Wake up service, user can click on
, the following window
will pop up:
Figure 180: Create New Wake Up Service
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Table 100: PMS Wake up Service
Field
Description
Room Number
Select the room number where to call
Time
Set the time of the wake up call
Action Status
Show the status of the call:

Programmed: the call is scheduled for the time
set
Type

Cancelled: the call is canceled

Executed: the wakeup call is made

Single: The call will be made once on the
specific time.

Daily: The call will be repeated every day on
the specific time
Once the call is made on the time specified, the following figure show the status of the wake up call.
Figure 181: Wake up Call executed
This call has been executed but has been rejected, that why we can see the “Busy” status.
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STATUS AND REPORTING
PBX Status
The UCM6510 monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces, Digital
Channels and Parking lot. It presents administrators the real time status in different sections under web
GUI->Status->PBX Status.
Figure 182: Status->PBX Status
Trunks
Users could see all the configured trunk status in this section.
Figure 183: Trunk Status
Table 101: Trunk Status
Display trunk status.
Status

Analog trunk/Digital trunk status:
Available
Busy
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Unavailable
Unknown Error
Error Configured: Incorrect signaling configuration between the two
devices. For example, both of the devices are configured as CPE or NET.

SIP Peer trunk status:
Unreachable: The hostname cannot be reached.
Unmonitored: QUALIFY feature is not turned on to be monitored.
Reachable: The hostname can be reached.

SIP Register trunk status:
Registered
Unrecognized Trunk
Display trunk name
Trunks
Display trunk Type:
Type

Analog

E1/T1/J1

SIP

IAX
Username
Display username for this trunk.
Port/Hostname/IP
Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.
Other operations are also available in trunk status section:

Click on "Trunks", the web page will redirect to trunk configuration page which can also be accessed
via web GUI->PBX->Basic/Call Routes->Analog Trunks.

Click on

Click on [ + ] to expand the status detail table.

Click on [ - ] to hide the status detail table.
to refresh the trunk status.
Extensions
Extensions in this section will be automatically sorted based on their status: idle, ringing, talking or
unavailable, and display them accordingly on the web UI status section.
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Figure 184: Extension Status
Table 102: Extension Status
Display extension number (including feature code). The color indicator has the
following definitions.
Status

Green: Free

Blue: Ringing

Yellow: In Use

Grey: Unavailable
Extension
Display the extension number.
Name/Label
Display name (callerID name) or label for the extension.
Display message status for the extension.
Message
Example: 2/4/1
Description: There are 2 urgent messages, 4 messages in total and 1 message
that has been already read.
Displays extension type.
Type

SIP User

IAX User

Analog User (FXS)

Features
Other operations are also available in extension status section:

Click on "Extensions", the web page will redirect to extension configuration page which can also be
accessed via web GUI->PBX->Basic/Call Routes->Extensions.

Click on
to refresh the extension status.
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
Click on one of the tabs
to display
the corresponding extensions accordingly.

Click on [ + ] to expand the status detail table.

Click on [ - ] to hide the status detail table.
Queues
Users could see all the configured call queue status in this section. The following figure shows the call
queue 6500 being in used.
Figure 185: Queue Status
The
current
call
status
(caller
ID,
duration),
agent
status,
service
level,
calls
summary
(completed/abandoned) are shown for the call queue. The agent status is defined as below.
Table 103: Agent Status
The agent is available/idle.
The agent is ringing.
The agent is talking/busy.
The agent has been logged out.
On the UCM6510, Service Level is defined as the percentage of high-quality calls over all calls in the call
queue, where high-quality call means calls answered within 10 seconds.
Other operations are also available in queue status section:

Click on "Queues", the web page will redirect to call queue configuration page which can also be
accessed via web GUI->PBX->Call Features->Call Queue.

Click on

Click on [ + ] to expand the call queue detail.

Click on [ - ] to hide the call queue detail.
to refresh the call queue status.
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Conference Rooms
Users could see all the conference room status in this section. It shows all the configured conference rooms,
current users, call duration for each user and conference call.
Figure 186: Conference Room Status
Other operations are also available in conference room status section:

Click on "Conference Rooms", the web page will redirect to conference room configuration page which
can also be accessed via web GUI->PBX->Call Features->Conference.

Click on

Click on [ + ] to expand the conference room details.

Click on [ - ]
to refresh the conference room status.
to hide the conference room details.
Interfaces Status
This section displays interface connection status on the UCM6510 for USB, SD Card, LAN, WAN, LAN
PoE, Heartbeat, Power 1, Power 2, Digital, FXS and FXO ports.
Table 104: Interface Status Indicators
FXO
Disconnected
Connected but not configured
Connected and idle
Connected and in use
FXS
Connected but not configured
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Connected and idle
Connected and in use
SD Card
SD Card plugged in
SD Card unplugged
USB
USB plugged in
USB unplugged
LAN PoE
PoE is used
PoE is not used
Power 1/2
Power supply is working
Power supply is abnormal
No power supply
LAN/WAN/Heart Beat
Connected
Not connected
Digital Port T1/E1/J1
Connected and working
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RED alarm: there is physical wiring problem,
loss of connectivity, or a framing/line-coding
mismatch with the remote switch.
YELLOW alarm: connected but the link is working
only one-way. This means that the remote switch
is not able to maintain sync with you, or is not
receiving your transmission.
The following example scenarios could trigger
YELLOW alarm:
1. The T1 port is connected with J1 connection.
2. Incorrect cable is used.
3. When using E1, one end is using CRC4 while
the other end is not.
BLUE alarm: the port goes into BLUE alarm
when it receives all unframed 1s on all timeslots
from the remote switch. This is a special signal
to indicate that the remote switch is having
problem with its upstream connection.
Cannot start up
Other operations are also available in interface status section:

Click on

Click on [ + ] to expand the interface details.

Click on [ - ] to hide the interface details.
to refresh the interface status.
Digital Channels Status
This section displays the status of the digital trunks on the UCM6510.
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Figure 187: Digital Channels Status
Table 105: Digital Channel Status Indicators
Voice Channel
Not available
Connected and in talking status; or incorrect
configuration
Connected and in idle status
Connected and in ringing status
Data Channel
Always shows blue square
D Channel
Always shows grey with channel number in blue.
Other operations are also available in interface status section:

Click on

Click on [ + ] to expand the Digital Channels Status details.

Click on [ - ] to hide the Digital Channels Status details.
to refresh the Digital Channels status.
Parking Lot
The UCM6510 supports call park using feature code. When there is call being parked, this section will
display the parking lot status.
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Figure 188: Parking Lot Status
Table 106: Parking Lot Status
Caller ID
Display the caller ID who parks the call.
Channel
Display channel for the call park.
Extension
Display the parking lot number where the call is parked/retrieved.
Display timeout (in seconds) for the parked call. The status page will dynamically
update this timer from 120 seconds (default) to 0. When the timer reaches 0, the
Timeout
caller who parks the call will be called back.
Other operations are also available in parking lot status section:

Click on "Parking Lot", the web page will redirect to feature codes page which can also be accessed
via web GUI->PBX->Internal Options->Feature Codes.

Click on

Click on [ + ] to expand the parking lot details.

Click on [ - ] to hide the parking details.
to refresh the parking lot status.
System Status
The UCM6510 system status can be accessed via web GUI->Status->System Status, which displays the
following system information.

General

Network

Storage Usage

Resource Usage
General
Under web GUI->Status->System Status->General, users could check the hardware and software
information for the UCM6510. Please see details in the following table.
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Table 107: System Status->General
Status ->System Status -> General
Model
Product model.
Part Number
Product part number.
System Time
Current system time. The current system time is also available on the upper right of
each web page.
Up Time
System up time since the last reboot.
Idle Time
System idle 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->Status->System Status->Network, users could check the network information for the
UCM6510. Please see details in the following table.
Table 108: System Status->Network
Status -> System Status -> Network
MAC Address
Global unique ID of device, in HEX format. The MAC address can be found on the label
coming with original box and on the label located on the bottom of the device.
IP Address
IP address.
Gateway
Default gateway address.
Subnet Mask
Subnet mask address.
DNS Server
DNS Server address.
Storage Usage
Users could access the storage usage information from web GUI->Status->System Status->Storage
Usage. It shows the available and used space for the following partitions.

Configuration partition
This partition contains PBX system configuration files and service configuration files.

Data partition
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Voicemail, recording files, IVR file, Music On Hold files and etc.

USB disk
USB disk will display if connected.

SD Card
SD Card will display if connected.
Figure 189: System Status->Storage Usage
Resource Usage
When configuring and managing the UCM6510, users could access resource usage information to estimate
the current usage and allocate the resources accordingly. Under web GUI->Status->System Status>Resource Usage, the current CPU usage and Memory usage are shown in the pie chart.
Figure 190: System Status->Resource Usage
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System Events
The UCM6510 can monitor important system events, log the alerts and send Email notifications to the
system administrator.
Alert Events List
The system alert events list can be found under web GUI->Status->System Events->Alert Events List.
The following event are currently supported on the UCM6510 which will have alert and/or Email generated
if occurred:
Disk Usage
External Disk Usage
Modify Admin Password
Memory Usage
System Reboot
System Update
System Crash
Register SIP Failed
Register SIP Trunk Failed
Restore Config
User Login Success
User Login Failed
SIP Internal Call Failure
SIP Outgoing Call through Trunk Failure
Fail2ban Blocking
SIP Lost Registration
SIP Peer Trunk Status
Click on
to configure the parameters for each event.
1. Disk Usage
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Figure 191: System Events->Alert Events Lists: Disk Usage

Detect Cycle: The UCM6510 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 UCM6510
system will send the alert.
2. External Disk Usage
Figure 192: System Events->Alert Events Lists: External Disk Usage

Detect Cycle: The UCM6510 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 UCM6510
system will send the alert.
3. Modify Admin Password
Figure 193: System Events->Alert Events Lists: Modify Admin Password
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
Detect Cycle: The UCM6510 will initiate the admin password check based on this cycle. Users can
enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
4. Memory Usage
Figure 194: System Events->Alert Events Lists: Memory Usage

Detect Cycle: The UCM6510 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 UCM6510
system will send the alert.
5. System Reboot
Figure 195: System Events->Alert Events Lists: System Reboot

Detect Cycle: The UCM6510 will check the system reboot based on this cycle. Users can enter
the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
6. System Update
Figure 196: System Events->Alert Events Lists: System Update

Detect Cycle: The UCM6510 will check the system update based on this cycle. Users can enter
the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
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7. System Crash
Figure 197: System Events->Alert Events Lists: System Crash

Detect Cycle: The UCM6510 will detect the event at each cycle based on the specified time.
Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the
cycle.
Click on the switch
to turn on/off the alert and Email notification for the event.
Users could also select the checkbox for each event and then click on button "Alert On", "Alert Off",
"Email Notification On", "Email Notification Off" to control the alert and Email notification configuration.
Alert Log
Under web GUI->Status->System Events->Alert Log, system messages are listed when the alert is
triggered for the configured system events. The following picture shows disk usage alert log. We can tell
the detect cycle for the disk usage is 10 minutes and the disk usage is restored to normal after the
administrator cleans up the disk storage below the threshold.
Figure 198: System Events->Alert Log
The following screenshot shows system crash alert logs.
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Figure 199: 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
into two types by the system:
. Alert logs are classified
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 200: Filter for Alert Log
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Alert Contact
Users could add administrator's Email address under web GUI->Status->System Events->Alert Contact
to send the alert notification to. Up to 10 Email addresses can be added.
CDR
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,
and etc.
On the UCM6510, the CDR can be accessed under web UI->Status->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.
Figure 201: CDR Filter
Table 109: CDR Filter Criteria
Inbound calls
Inbound calls are calls originated from a non-internal source (like a VoIP trunk) and
sent to an internal extension.
Outbound calls
Outbound calls are calls sent to a non-internal source (like a VoIP trunk) from an
internal extension.
Internal calls
Internal calls are calls from one internal extension to another extension, which are not
sent over a trunk.
External calls
External calls are calls sent from one trunk to another trunk, which are not sent to any
internal extension.
Inbound Trunks
Select certain inbound trunk(s) and the CDR of calls going inbound through the trunk(s)
will be filtered out.
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Outbound Trunks
Select certain outbound trunk(s) and the CDR of calls going outbound through the
trunk(s) will be filtered out.
Caller Number
Enter the caller number to filter the CDR report. CDR with the matching caller number
will be filtered out.
User could specify a particular caller number or enter a pattern. ‘.’ matches zero or
more characters, only appears in the end. ‘X’ matches any digit from 0 to 9, caseinsensitive, repeatable, only appears in the end.
For example:
3XXX: It will filter out CDR that having caller number with leading digit 3 and of 4 digits’
length.
3.: It will filter out CDR that having caller number with leading digit 3 and of any length.
Caller Name
Enter the caller name to filter the CDR report. CDR with the matching caller name will
be filtered out.
Callee Number
Enter the callee number to filter the CDR report. CDR with the matching callee number
will be filtered out.
Start Time
Specify the start time to filter the CDR report. Click on the calendar icon on the right
and the calendar will show for users to select the exact date and time.
End Time
Specify the end time to filter the CDR report. Click on the calendar icon on the right
and the calendar will show for users to select the exact date and time.
The call report will display as the following figure shows.
Figure 202: Call Report
The CDR report has the following data fields:

Start Time
Format: 2016-06-30 06:10:45

Call Type
Example:
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IVR
DIAL
WAKEUP

Call From
Example format:
"Mhammed elmrabet" 2009

Call To
Example format:
2002

Call Time
Format: 0:00:45

Talk Time
Format: 0:00:40

Account Code
Example format:
Test/Grandstream

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 Searched Results
Click on “Download Search Result(s)” to export the records filtered out to a .csv file.

Download All Records
Click on “Download All Records” to export all the records to a .csv file.

Delete All
On the bottom of the page, click on "Delete All" button to remove all the call report information.

Play/Download/Delete Recording File (per entry)
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If the entry has audio recording file for the call, the three icons on the most right column will be
activated for users to select. In the following picture, the second entry has audio recording file for
the call.
Click on
to play the recording file; click on
to download the recording file in .wav format;
click on
to delete the recording file (the call record entry will not be deleted).
Figure 203: Call Report Entry with Audio Recording File

Automatic Download CDR Records
User could configure the UCM6510 to automatically download the CDR records and send the records
to an Email address. Click on “Automatic Download Settings”, and configure the parameters in the
dialog below.
Figure 204: Automatic Download Settings
To receive CDR record automatically from Email, check “Enable” and select a time period “By Day” “By
Week” or “By Month”, select Hour of the day as well for the automatic download period. Make sure you
have entered an Email or multiple email addresses where to receive the CDR records.
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CDR Improvement
Starting from UCM6510 firmware 1.0.10.x, transferred call will no longer be displayed as a separate call entry in
CDR. It will display within call record in the same entry. CDR new features can be found under web UI-> Status>CDR->CDR. The user can click the option field to view call history, such as premier caller and transferred call
information.
Figure 205: CDR Report
Figure 206: Detailed CDR Information
Downloaded CDR File
The downloaded CDR (.csv file) has different format from the web UI 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 207: 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.
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from-internal: internal extension makes outbound calls.
ext-did-XXXXX: inbound calls. It starts with "ext-did", and "XXXXX" content varies case by case, which
also relate to the order when the trunk is created.
ext-local: internal calls between local extensions.

Source Channel, Dest Channel
Sample 1:
Figure 208: Downloaded CDR File Sample - Source Channel and Dest Channel 1
DAHDI means it is an analog call, FXO or FXS.
For UCM6510, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports.
Sample 2:
Figure 209: Downloaded CDR File Sample - Source Channel and Dest Channel 2
"SIP" means it's a SIP call. There are three possible format:
(a) PJSIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random
string and can be ignored.
(c) PJSIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out
through the trunk.
(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
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Statistics
CDR Statistics is an additional feature on the UCM6510 which provides users a visual overview of the call
report across the time frame. Users can filter with different criteria to generate the statistics chart.
Figure 210: CDR Statistics
Table 110: CDR Statistics Filter Criteria
Trunk Type
Call Type
Time Range
Select one of the following trunk type.

All

SIP Calls

PSTN Calls
Select one or more in the following checkboxes.

Inbound calls

Outbound calls

Internal calls

External calls

All calls

By month (of the selected year).

By week (of the selected year).

By day (of the specified month for the year).

By hour (of the specified date).

By range. For example, 2013-01 To 2013-03.
Recording Files
This page lists all the recording files recorded by "Auto Record" per extension/ring group/call queue/trunk,
or via feature code "Audio Mix Record". If external storage device is plugged in, for example, SD card or
USB drive, the files are stored on the external storage. Otherwise, internal storage will be used on the
UCM6510.
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Figure 211: CDR->Recording Files

Click on “Delete Selected Recording Files” to delete the recording files.

Click on “Delete All Recording Files” to delete all recording files.

Click on
to download the recording file in .wav format.

Click on
to delete the recording file.

To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding column.
Click on the title again can switch the sorting mode between ascending order or descending order.
API Configuration Files
The UCM6510 supports third party billing interface API for external billing software to access CDR and call
recordings on the PBX. The API uses HTTPS to request the CDR data and call recording data matching
given parameters as configured on the third party application.
Before accessing the API, the administrators need enable API and configure the access/authentication
information on the UCM6510 first. The API configuration parameters are listed in the table below.
Table 111: CDR API Configuration Files
Enable
Enable/Disable CDR API. The default setting is disabled.
TLS Bind Address
Configure the IP address for TLS server to bind to. "0.0.0.0" means binding to all
interfaces. The port number is optional and the default port number is 8443. The IP
address must match the common name (host name) in the certificate so that the TLS
socket won't bind to multiple IP addresses. The default setting is 0.0.0.0:8443.
TLS Private Key
Upload TLS private key. The size of the key file must be under 2MB. This file will be
renamed as 'private.pem' automatically.
TLS Cert
Upload TLS cert. The size of the certificate must be under 2MB. This is the certificate
file (*.pem format only) for TLS connection. This file will be renamed as
"certificate.pem" automatically. It contains private key for the client and signed
certificate for the server.
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Username
Configure the Username for API Authentication.
Password
Configure the Password for API Authentication.
Permitted
Specify a list of IP addresses permitted by API. This creates an IP-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.
The format of the HTTPS request for the CDR API is as below.
https://[UCM IP]:[Port]/cdrapi?[option1]=[value]&[option2]=[value]&...
By default, the port number for the API is 8443.
The options included in the request URI control the record matching and output format. For CDR
matching parameters, all non-empty parameters must have a match to return a record. Parameters can
appear in the URI in any order. Multiple values given for caller or callee will be concatenated. The
following table shows the parameter list used in the CDR API.
Table 112: CDR API URI Parameters
Field
Value
format
csv, xml, json
Details
Define the format for output of matching CDR rows.
Default is csv (comma separated values).
Number of records to return. Default is 1000, which is
numRecords
Number: 0-1000
also the maximum allowed value.
Number of matching records to skip. This will be
offset
combined with numRecords to receive all matches over
Number
multiple responses. Default is 0.
caller
callee
Comma separated extensions,
Filters based on src (caller) or dst (callee) value,
ranges of extensions, or
matching any extension contained in the parameter input
regular expressions.
string.
Example:
Patterns containing one or more wildcards ('@' or '_') will
match as a regular expression, and treat '-' as a literal
caller=5300,5302-5304,_4@
-OR-
hyphen rather than a range signifier. The '@' wildcard
matches any number of characters (including zero),
while '_' matches any single character. Otherwise,
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caller=5300&caller=5302-
patterns containing a single hyphen will be matching a
5304&caller=_4@
range of numerical extensions, with non-numerical
(Matches extensions 5300,
5302, 5303, 5304, and any
extension containing 4 as the
characters ignored, while patterns containing multiple
hyphens will be ignored. (The pattern "0-0" will match all
non-numerical and empty strings).
second digit/character).
Date and/or time of day in any
startTime
of the following formats:
YYYY-MM-DDTHH:MM
YYYY-MM-DDTHH:MM:SS
Filters based on the start (call start time) value. Calls
which start within this period (inclusive of boundaries)
will match, regardless of the call answer or end time. An
endTime
YYYY-MM-
empty value for either field will be interpreted as range
DDTHH:MM:SS.SSS
with no minimum or maximum respectively.
(literal 'T' character separator
Strings without a date have a default value of 2000-01-
in above three formats)
01. Strings without a time of day have a default value of
of 00:00 UTC, while strings with a time of day specified
HH:MM
may also optionally specify a time zone offset - replace
HH:MM:SS
'+' in time zone offset with '%2B' (see
http://www.w3.org/TR/NOTE-datetime).
HH:MM:SS.SSS
now
DDDDDDDDDD
minDur
Filters based on the billsec value, the duration between
Number (duration in seconds)
call answer and call end.
maxDur
Example Queries:
The following illustrates the format of queries to accomplish certain requests. In most cases, multiple
different queries will accomplish the same goal, and these examples are not intended to be exhaustive, but
rather to bring attention to particular features of the CDR API connector.
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Query 1: Request all records of calls placed on extension 5300 which last between 8 and 60 seconds
(inclusive), with results in CSV format.
https://192.168.254.200:8088/cdrapi?format=CSV&caller=5300&minDur=8&maxDur=60
-ORhttps://192.168.254.200:8088/cdrapi?caller=5300&minDur=8&maxDur=60
Query 2: Request all records of calls placed on extension 5300 or in the range 6300-6399 to extensions
starting with 5, with results in XML format.
https://192.168.254.200:8088/cdrapi?format=XML&caller=5300,6300-6399&callee=5@
-ORhttps://192.168.254.200:8088/cdrapi?cdrapi?format=XML&caller=5300&caller=6300-6399&callee=5@
Query 3: Request all records of calls placed on extensions containing substring "53" prior to January 23,
2013 00:00:00 UTC to extensions 5300-5309, with results in CSV format.
https://192.168.254.200:8088/cdrapi?caller=@53@&callee=5300-5309&endTime=2013-01-23
-ORhttps://192.168.254.200:8088/cdrapi?caller=@53@&callee=530_&endTime=2013-01-23T00:00:00
Query 4: Request all records of calls placed by an Anonymous caller during July 2013 Central Standard
Time to extensions starting with 2 or 34 or ending with 5, with results in CSV format.
https://192.168.254.200:8088/cdrapi?caller=Anonymous&callee=2@,34@,@5&startTime=2013-07-01T00:00:0006:00&endTime=2013-07-31T23:59:59-06:00
Query 5: Request all records during July 2013 Central Standard Time, 200 at a time, with results in CSV
format.
https://192.168.254.200:8088/cdrapi?startTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:5906:00&numRecords=200&offset=0
-THENhttps://192.168.254.200:8088/cdrapi?sstartTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:59-
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06:00&numRecords=200&offset=200
-THENhttps://192.168.254.200:8088/cdrapi?startTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:5906:00&numRecords=200&offset=400
--------------------------------------------------------------------------------------------------------------------- ----------------------Note:

Disallowed characters in the caller, callee, startTime, or endTime strings, and non-digit characters
in the values of numRecords, offset, minDur, or maxDur, will result in no records returned - the
appropriate container/header for the output format will be the only output. If the format parameter
is in error, the CSV header will be used. Error messages will appear in the Asterisk log (along with
errors stemming from failed database connections, etc.).

Other errors which return no records include:
- Multiple hyphens in an extension range (e.g. caller=5300-5301-,6300)
- Empty parameter value (e.g. caller=)
- Extension values starting with comma, or with consecutive commas (e.g. caller=5300,,5303)
- Unknown parameters (e.g. caler=5300) or URI ending with '&'
- Except for caller and callee, multiple instances of the same parameter within the URI (e.g.
minDur=5&minDur=10)
--------------------------------------------------------------------------------------------------------------------- -----------------------
Example Output:
The following are examples of each of the output formats for the same data set.
CSV:
AcctId,accountcode,src,dst,dcontext,clid,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,
billsec,disposition,amaflags,uniqueid,userfield,channel_ext,dstchannel_ext,service
62,,5300,5301,from-internal,"pn01" <5300>,SIP/5300-00000000,SIP/530100000001,Dial,SIP/5301,60,,2013-12-03 11:46:40,2013-12-03 11:46:43,2013-12-03
11:46:49,9,6,ANSWERED,DOCUMENTATION,1386092800.0,EXT,5300,5301,s
63,,5300,5301,from-internal,"pn01" <5300>,SIP/5300-00000000,SIP/530100000001,Dial,SIP/5301,60,,2013-12-03 14:01:41,2013-12-03 14:01:43,2013-12-03
14:01:46,5,3,ANSWERED,DOCUMENTATION,1386100901.0,EXT,5300,5301,s
64,,5300,5301,from-internal,"pn01" <5300>,SIP/5300-00000002,SIP/530100000003,Dial,SIP/5301,60,,2013-12-03 14:02:23,2013-12-03 14:02:27,2013-12-03
14:02:31,8,4,ANSWERED,DOCUMENTATION,1386100943.2,EXT,5300,5301,s
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XML:
<root>
<cdr><AcctId>62</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext
>from-internal</dcontext><clid>&quot;pn01&quot; &lt;5300&gt;</clid><channel>SIP/530000000000</channel><dstchannel>SIP/530100000001</dstchannel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03
11:46:40</start><answer>2013-12-03 11:46:43</answer><end>2013-12-03
11:46:49</end><duration>9</duration><billsec>6</billsec><disposition>ANSWERED</disposition><a
maflags>DOCUMENTATION</amaflags><uniqueid>1386092800.0</uniqueid><userfield>EXT</userfie
ld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service></
cdr>
<cdr><AcctId>63</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext
>from-internal</dcontext><clid>&quot;pn01&quot; &lt;5300&gt;</clid><channel>SIP/530000000000</channel><dstchannel>SIP/530100000001</dstchannel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03
14:01:41</start><answer>2013-12-03 14:01:43</answer><end>2013-12-03
14:01:46</end><duration>5</duration><billsec>3</billsec><disposition>ANSWERED</disposition><a
maflags>DOCUMENTATION</amaflags><uniqueid>1386100901.0</uniqueid><userfield>EXT</userfie
ld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service></
cdr>
<cdr><AcctId>64</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext
>from-internal</dcontext><clid>&quot;pn01&quot; &lt;5300&gt;</clid><channel>SIP/530000000002</channel><dstchannel>SIP/530100000003</dstchannel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03
14:02:23</start><answer>2013-12-03 14:02:27</answer><end>2013-12-03
14:02:31</end><duration>8</duration><billsec>4</billsec><disposition>ANSWERED</disposition><a
maflags>DOCUMENTATION</amaflags><uniqueid>1386100943.2</uniqueid><userfield>EXT</userfie
ld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service></
cdr>
</root>
JSON:
{
"cdr":
[
{ "AcctId": "62", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal",
"clid": "\"pn01\" <5300>", "channel": "SIP/5300-00000000", "dstchannel": "SIP/5301-00000001",
"lastapp": "Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 11:46:40", "answer": "2013-12-03
11:46:43", "end": "2013-12-03 11:46:49", "duration": "9", "billsec": "6", "disposition": "ANSWERED",
"amaflags": "DOCUMENTATION", "uniqueid": "1386092800.0", "userfield": "EXT", "channel_ext":
"5300", "dstchannel_ext": "5301", "service": "s" },
{ "AcctId": "63", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal",
"clid": "\"pn01\" <5300>", "channel": "SIP/5300-00000000", "dstchannel": "SIP/5301-00000001",
"lastapp": "Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 14:01:41", "answer": "2013-12-03
14:01:43", "end": "2013-12-03 14:01:46", "duration": "5", "billsec": "3", "disposition": "ANSWERED",
"amaflags": "DOCUMENTATION", "uniqueid": "1386100901.0", "userfield": "EXT", "channel_ext":
"5300", "dstchannel_ext": "5301", "service": "s" },
{ "AcctId": "64", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal",
"clid": "\"pn01\" <5300>", "channel": "SIP/5300-00000002", "dstchannel": "SIP/5301-00000003",
"lastapp": "Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 14:02:23", "answer": "2013-12-03
14:02:27", "end": "2013-12-03 14:02:31", "duration": "8", "billsec": "4", "disposition": "ANSWERED",
"amaflags": "DOCUMENTATION", "uniqueid": "1386100943.2", "userfield": "EXT", "channel_ext":
"5300", "dstchannel_ext": "5301", "service": "s" }
]
}
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UPGRADING AND MAINTENANCE
Upgrading
The UCM6510 can be upgraded to a new firmware version remotely or locally. This section describes how
to upgrade your UCM6510 via network or local upload.
Upgrading via Network
The UCM6510 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 212: Network Upgrade
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Table 113: 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 UCM6510.
Firmware File Suffix
If configured, only the firmware with the matching encrypted postfix will be
downloaded and flashed into the UCM6510.
HTTP/HTTPS User Name
The user name for the HTTP/HTTPS server.
HTTP/HTTPS Password
The password for the HTTP/HTTPS server.
Please follow the steps below to upgrade the firmware remotely.

Enter the firmware server path under web GUI->Maintenance->Upgrade.

Click on "Save". Then reboot the device to start the upgrading process.

Please be patient during the upgrading process. Once done, a reboot message will be displayed in the
LCD.

Manually reboot the UCM6510 when it's appropriate to avoid immediate service interruption. After it
boots up, log in the web GUI to check the firmware version.
Upgrading via Local Upload
If there is no HTTP/TFTP server, users could also upload the firmware to the UCM6510 directly via web
GUI. Please follow the steps below to upload firmware locally.
1. Download the latest UCM6510 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
and select the
firmware file from your PC. The default firmware file name is ucm6510fw.bin
Figure 213: Local Upgrade
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4. Click on
to start upgrading.
Figure 214: Upgrading Firmware Files
5. Wait until the upgrading process is successful and a window will be popped up in the web GUI.
Figure 215: Reboot UCM6510
6. Click on "OK" to reboot the UCM6510 and check the firmware version after it boots up.
--------------------------------------------------------------------------------------------------------------------- -----------------------
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Note:
Please do not interrupt or power cycle the UCM6510 during upgrading process.
--------------------------------------------------------------------------------------------------------------------- -----------------------
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 UCM6510 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 UCM6510 web configuration interface;
5. Configure the Firmware Server Path to the IP address of the PC;
6. Update the changes and reboot the UCM6510.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use
Microsoft IIS web server.
Backup
The UCM6510 configuration can be backed up locally or via network. The backup file will be used to restore
the configuration on UCM6510 when necessary.
Backup/Restore
Users could backup the UCM6510 configurations for restore purpose under web GUI->Maintenance>Backup->Backup / Restore. Click on
to create a new backup. Then the following
dialog will show.
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Figure 216: Create New Backup
1. Choose the files to be included in the backup.
2. Choose where to store the backup file: USB Disk, SD Card or Local.
3. Name the backup file.
4. Click on "Backup" to start backup.
Once the backup is done, the list of the backups will be displayed with date and time in the web page. Users
can download
, restore
, or delete
it from the UCM6510 internal storage or the external device.
Click on
to upload backup file from the local device to UCM6510. The uploaded backup
file will also be displayed in the web page and can be used to restore the UCM6510.
Figure 217: Backup / Restore
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option allows UCM6510 to perform automatically backup on the user specified time.
Regular backup file can only be stored in USB / SD card / SFTP server. User is allowed to set backup time
from 0-23 and how frequent the backup will be performed.
Figure 218: Local Backup
Data Sync
Besides local backup, users could backup the voice records/voice mails/CDR/FAX in a daily basis to a
remote server via SFTP protocol automatically under Web GUI->Maintenance->Backup->Data Sync.
The client account supports special characters such as @ or “.”. This change allows user to use email
address as SFTP accounts. It allows users as well to specify the destination directory on SFTP server for
backup file. If the directory doesn’t exist on the destination, UCM6510 will create the directory automatically.
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Figure 219: Data Sync
Table 114: Data Sync Configuration
Enable Data Sync
Enable the auto backup 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 could click on "Test Connection". The UCM6510 will then try
connecting the server to make sure the server is up and accessible for the UCM6510.
Save the changes and all the backup logs will be listed on the web page.
Restore Configuration from Backup File
To restore the configuration on the UCM6510 from a backup file, users could go to web GUI->Maintenance>Backup->Backup / Restore.

A list of previous configuration backups is displayed on the web page. Users could click on
of the
desired backup file and it will be restored to the UCM6510.
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
If users have other backup files on PC to restore on the UCM6510, click on "Upload Backup File" first
and select it from local PC to upload on the UCM6510. 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.

User could also restore using the backup file saved in SD card or USB device plugged into the
UCM6510.
Figure 220: Restore UCM6510 from Backup File
----------------------------------------------------------------------------------------------------------------------------- --------------Note:

The uploaded backup file must be a tar file with no special characters like *,!,#,@,&,$,%,^,(,),/,\,space
in the file name.

The uploaded back file size must be under 10MB.
----------------------------------------------------------------------------------------------------------------------------- ---------------
Cleaner
Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under
web GUI->Maintenance->Cleaner.
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Figure 221: Cleaner
Table 115: Cleaner Configuration
Enable CDR Cleaner
Enable the CDR Cleaner function.
CDR Clean Time
Enter 0-23 to specify the hour of the day to clean up CDR.
Clean Interval
Enter 1-30 to specify the day of the month to clean up CDR.
Enable VR Cleaner
Enter the Voice Records Cleaner function.
Choose Cleaner File
Select the file type(s) to be cleaned automatically.
VR Clean Threshold
Specify the Voice Records threshold from 0 to 99 by using local storage
status in percentage.
VR Clean Time
Enter 0-23 to specify the hour of the day to clean up Voice Records.
Clean Interval
Enter 1-30 to specify the day of the month to clean up Voice Records.
All the cleaner logs will be listed on the bottom of the page.
Reset and Reboot
Users could perform reset and reboot under web GUI->Maintenance->Reset and Reboot.
To factory reset the device, select the mode type first. There are two different types for reset.

User Data: All the data including voicemail, recordings, IVR Prompt, Music on Hold, CDR and backup
files will be cleared.
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
All: All the configurations and data will be reset to factory default.
Figure 222: Reset and Reboot
Syslog
On the UCM6510, users could dump the syslog information to a remote server under web GUI>Maintenance->Syslog. Enter the syslog server hostname or IP address and select the module/level for
the syslog information.
The default syslog level for all modules is "error", which is recommended in your UCM6510 settings
because it can be helpful to locate the issues when errors happen.
Some typical modules for UCM6510 functions are as follows and users can turn on "notic" and "verb" levels
besides "error" level.
pbx: This module is related to general PBX functions.
chan_sip: This module is related to SIP calls.
chan_dahdi: This module is related to analog calls (FXO/FXS).
app_meetme: This module is related to Conference Bridge.
--------------------------------------------------------------------------------------------------------------------- ----------------------Note:
Syslog is usually for debugging and troubleshooting purpose. Turning on all levels for all syslog modules is
not recommended for daily usage. Too many syslog print might cause traffic and affect system performance.
--------------------------------------------------------------------------------------------- -----------------------------------------------
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Troubleshooting
On the UCM6510, users could capture traces, ping remote host and traceroute remote host for
troubleshooting purpose under web GUI->Maintenance->Troubleshooting.
Ethernet Capture
The captured trace can be downloaded for analysis. Also the instructions or result will be displayed in the
web GUI output result.
Figure 223: Ethernet Capture
The output result is in .pcap format. Therefore, users could specify the capture filter as used in general
network traffic capture tool (host, src, dst, net, protocol, port, port range) before starting capturing the trace.
IP Ping
Enter the target host in host name or IP address. Then press "Start" button. The output result will
dynamically display in the window below.
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Figure 224: PING
Traceroute
Enter the target host in host name or IP address. Then press "Start" button. The output result will
dynamically display in the window below.
Figure 225: Traceroute
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PRI/SS7/MFC/R2 Signaling Trace
Please see section [Digital Trunk Troubleshooting].
Analog Record Trace
Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM6510 user has
caller ID issue for incoming call from Analog trunk. Users can access analog record trance under web GUI>Maintenance->Troubleshooting ->Analog Record Trace.
Here is the step to capture trace:
1. Select FXO or FXS for "Record Ports". If the issue happens on FXO 1, select FXO port 1 to record
the trace.
2. Select "Record Direction".
3. Select "Record File Mode" to separate the record per direction or mix.
4. Click on "Start".
5. Make a call via the analog port that has the issue.
6. Once done, click on "Stop".
7. Click on "Download" to download the analog record trace.
Figure 226: Troubleshooting Analog Trunks
After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream
Technical support in the following link for further assistance if the issue is not resolved.
http://www.grandstream.com/support
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E&M Immediate Record Trace
Before capturing the trace for E&M, user could configure “Record Direction” and “Record File Mode".
“Record File Mode” can be separate (one record per direction) or mix. Then click on “Start” to start to
capture.
Figure 227: E&M Immediate Record Trace
Service Check
Enable Service Check to periodically check UCM6510. Check Cycle is configurable in seconds and the
default setting is 60 sec. Check Times is the maximum number of failed checks before restart the UCM6510.
The default setting is 3. If there is no response from UCM6510 after 3 attempts (default) to check, current
status will be stored and UCM6510 will be restarted.
Figure 228: Service Check
Network Status
UCM6510 supports Network Status to display active internet connections (Severs and Established). User can
use Network Status to troubleshoot connection issue between UCM6510 and other services.
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Figure 229: Network Status
Remote Access
SSH Access
SSH switch now is available via web UI and LCD. User can enable or disable SSH access directly from
web UI or LCD screen. For web SSH access, please log in UCM6510 web interface and go to Maintenance>Remote Access->SSH Access. By default, SSH access is disabled for security concerns. It is highly
recommended to only enable SSH access for debugging purpose.
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Figure 230: SSH Access
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EXPERIENCING THE UCM6510 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 UCM6510 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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