UCM6100 User Manual

Grandstream Networks, Inc.
UCM6100 Series IP PBX
User Manual
This page intentionally left blank
UCM6100 Series IP PBX User Manual
Table of Content
GNU GPL INFORMATION .......................................................................... 16
CHANGE LOG ............................................................................................. 17
FIRMWARE VERSION 1.0.8.11 ........................................................................................................ 17
FIRMWARE VERSION 1.0.7.11 ........................................................................................................ 18
FIRMWARE VERSION 1.0.6.11 ........................................................................................................ 18
FIRMWARE VERSION 1.0.6.10 ........................................................................................................ 18
FIRMWARE VERSION 1.0.5.19 ........................................................................................................ 19
FIRMWARE VERSION 1.0.5.14 ........................................................................................................ 19
FIRMWARE VERSION 1.0.4.7 .......................................................................................................... 20
FIRMWARE VERSION 1.0.3.13 ........................................................................................................ 20
FIRMWARE VERSION 1.0.2.21 ........................................................................................................ 21
FIRMWARE VERSION 1.0.1.22 ........................................................................................................ 21
WELCOME .................................................................................................. 23
PRODUCT OVERVIEW............................................................................... 25
FEATURE HIGHTLIGHTS ................................................................................................................. 25
TECHNICAL SPECIFICATIONS ........................................................................................................ 26
INSTALLATION ........................................................................................... 29
EQUIPMENT PACKAGING ............................................................................................................... 29
CONNECT YOUR UCM6100 ............................................................................................................ 29
CONNECT THE UCM6102 ......................................................................................................... 29
CONNECT THE UCM6104 ......................................................................................................... 31
CONNECT THE UCM6108 ......................................................................................................... 32
CONNECT THE UCM6116 ......................................................................................................... 33
SAFETY COMPLIANCES.................................................................................................................. 33
WARRANTY ...................................................................................................................................... 34
GETTING STARTED ................................................................................... 35
USE THE LCD MENU ....................................................................................................................... 35
USE THE LED INDICATORS ............................................................................................................ 37
USE THE WEB GUI .......................................................................................................................... 38
ACCESS WEB GUI .................................................................................................................... 38
WEB GUI CONFIGURATIONS ................................................................................................... 39
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WEB GUI LANGUAGES ............................................................................................................. 39
SAVE AND APPLY CHANGES ................................................................................................... 40
MAKE YOUR FIRST CALL ................................................................................................................ 40
SYSTEM SETTINGS ................................................................................... 41
USER MANAGEMENT ...................................................................................................................... 41
USER PRIVILEGES ................................................................................................................... 41
CREATE NEW WEB UI USER ................................................................................................... 42
CONCURRENT MULTI-USER LOGIN........................................................................................ 44
OPERATION LOG ...................................................................................................................... 44
NETWORK SETTINGS ..................................................................................................................... 46
BASIC SETTINGS ...................................................................................................................... 46
802.1X ........................................................................................................................................ 52
STATIC ROUTES ....................................................................................................................... 54
PORT FORWORDING (UCM6102 ONLY) .................................................................................. 56
FIREWALL......................................................................................................................................... 58
STATIC DEFENSE...................................................................................................................... 58
DYNAMIC DEFENSE ................................................................................................................. 61
FAIL2BAN ................................................................................................................................... 62
CHANGE PASSWORD ..................................................................................................................... 63
LDAP SERVER ................................................................................................................................. 63
LDAP SERVER CONFIGURATIONS .......................................................................................... 64
LDAP PHONEBOOK .................................................................................................................. 65
LDAP CLIENT CONFIGURATIONS ............................................................................................ 69
HTTP SERVER ................................................................................................................................. 70
EMAIL SETTINGS ............................................................................................................................. 71
TIME SETTINGS ............................................................................................................................... 72
AUTO TIME UPDATING ............................................................................................................. 72
SET TIME MANUALLY ............................................................................................................... 74
OFFICE TIME ............................................................................................................................. 74
HOLIDAY .................................................................................................................................... 76
NTP SERVER .................................................................................................................................... 78
RECORDINGS STORAGE ................................................................................................................ 79
LOGIN TIMEOUT SETTINGS ........................................................................................................... 81
PROVISIONING........................................................................................... 83
OVERVIEW ....................................................................................................................................... 83
CONFIGURATION ARCHITECTURE FOR END POINT DEVICE ..................................................... 83
AUTO PROVISIONING SETTINGS .................................................................................................. 84
DISCOVERY ..................................................................................................................................... 87
GLOBAL CONFIGURATION ............................................................................................................. 88
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GLOBAL POLICY ....................................................................................................................... 88
GLOBAL TEMPLATES ............................................................................................................... 95
MODEL CONFIGURATION ............................................................................................................... 97
MODEL TEMPLATES ................................................................................................................. 97
MODEL UPDATE ...................................................................................................................... 100
DEVICE CONFIGURATION ............................................................................................................ 100
CREATE NEW DEVICE ............................................................................................................ 100
MANAGE DEVICES ................................................................................................................. 101
SAMPLE APPLICATION .................................................................................................................. 108
EXTENSIONS .............................................................................................113
CREATE NEW USER .......................................................................................................................113
CREATE NEW SIP EXTENSION ...............................................................................................113
CREATE NEW IAX EXTENSION ...............................................................................................119
CREATE NEW FXS EXTENSION ............................................................................................ 123
BATCH ADD EXTENSIONS ............................................................................................................ 127
BATCH ADD SIP EXTENSIONS ............................................................................................... 127
BATCH ADD IAX EXTENSIONS............................................................................................... 130
EDIT EXTENSION ........................................................................................................................... 132
EXPORT EXTENSIONS .................................................................................................................. 133
IMPORT EXTENSIONS................................................................................................................... 134
EMAIL TO USER ............................................................................................................................. 135
TRUNKS .................................................................................................... 137
ANALOG TRUNKS .......................................................................................................................... 137
ANALOG TRUNK CONFIGURATION ....................................................................................... 137
PSTN DETECTION .................................................................................................................. 140
VOIP TRUNKS ................................................................................................................................ 143
Direct Outward Dialing (DOD)................................................................................................... 152
SLA STATION ............................................................................................ 155
CREATE/EDIT SLA STATION.......................................................................................................... 155
SAMPLE CONFIGURATION ........................................................................................................... 156
CALL ROUTES ......................................................................................... 159
OUTBOUND ROUTES .................................................................................................................... 159
INBOUND ROUTES ........................................................................................................................ 161
INBOUND RULE CONFIGURATIONS ..................................................................................... 162
BLACKLIST CONFIGURATIONS ............................................................................................. 163
CONFERENCE BRIDGE .......................................................................... 165
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CONFERENCE BRIDGE CONFIGURATIONS ......................................................................... 165
JOIN A CONFERENCE CALL .................................................................................................. 167
INVITE OTHER PARTIES TO JOIN CONFERENCE ................................................................ 167
DURING THE CONFERENCE ................................................................................................. 168
RECORD CONFERENCE ........................................................................................................ 169
IVR ............................................................................................................. 171
CONFIGURE IVR ............................................................................................................................ 171
CREATE CUSTOM PROMPT ......................................................................................................... 173
RECORD NEW CUSTOM PROMPT ........................................................................................ 173
UPLOAD CUSTOM PROMPT .................................................................................................. 174
LANGUAGE SETTINGS FOR VOICE PROMPT ..................................... 175
DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE.............................................................. 175
VOICEMAIL ............................................................................................... 177
CONFIGURE VOICEMAIL .............................................................................................................. 177
ACCESS VOICEMAIL ..................................................................................................................... 178
VOICEMAIL EMAIL SETTINGS....................................................................................................... 179
CONFIGURE VOICEMAIL GROUP ................................................................................................ 180
RING GROUP ............................................................................................ 183
CONFIGURE RING GROUP ........................................................................................................... 183
PAGING AND INTERCOM GROUP ......................................................... 185
CONFIGURE PAGING/INTERCOM GROUP .................................................................................. 185
CALL QUEUE ............................................................................................ 187
CONFIGURE CALL QUEUE ........................................................................................................... 187
EXTENSION GROUPS ............................................................................. 191
CONFIGURE EXTENSION GROUPS ............................................................................................. 191
USING EXTENSION GROUPS ....................................................................................................... 192
PICKUP GROUPS ..................................................................................... 193
CONFIGURE PICKUP GROUPS .................................................................................................... 193
CONFIGURE PICKUP FEATURE CODE ........................................................................................ 193
MUSIC ON HOLD ...................................................................................... 195
FAX/T.38 .................................................................................................... 197
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CONFIGURE FAX/T.38 ................................................................................................................... 197
SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE .............................................. 198
SAMPLE CONFIGURATION FOR FAX-TO-EMAIL ......................................................................... 200
ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS) ............ 201
BUSY CAMP-ON ....................................................................................... 203
FOLLOW ME ............................................................................................. 205
ONE-KEY DIAL ......................................................................................... 207
DISA ........................................................................................................... 209
CALLBACK FEATURE ..............................................................................211
BLF AND EVENT LIST ............................................................................. 213
BLF .................................................................................................................................................. 213
EVENT LIST .................................................................................................................................... 213
DIAL BY NAME ......................................................................................... 215
DIAL BY NAME CONFIGURATION ................................................................................................. 215
ACTIVE CALLS AND MONITOR.............................................................. 219
ACTIVE CALLS STATUS................................................................................................................. 219
HANG UP ACTIVE CALLS .............................................................................................................. 220
CALL MONITOR .............................................................................................................................. 220
CALL FEATURES ..................................................................................... 223
FEATURE CODES .......................................................................................................................... 223
CALL RECORDING ......................................................................................................................... 226
CALL PARK ..................................................................................................................................... 227
PARK A CALL ........................................................................................................................... 227
RETRIEVE THE PARKED CALL .............................................................................................. 228
INTERNAL OPTIONS ............................................................................... 229
INTERNAL OPTIONS/GENERAL .................................................................................................... 229
INTERNAL OPTIONS/JITTER BUFFER ......................................................................................... 231
INTERNAL OPTIONS/RTP SETTINGS ........................................................................................... 231
INTERNAL OPTIONS/PORTS CONFIG.......................................................................................... 232
INTERNAL OPTIONS/STUN MONITOR ......................................................................................... 233
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INTERNAL OPTIONS/PAYLOAD .................................................................................................... 234
IAX SETTINGS .......................................................................................... 235
IAX SETTINGS/GENERAL .............................................................................................................. 235
IAX SETTINGS/REGISTRATION .................................................................................................... 235
IAX SETTINGS/STATIC DEFENSE ................................................................................................. 236
SIP SETTINGS .......................................................................................... 237
SIP SETTINGS/GENERAL .............................................................................................................. 237
SIP SETTINGS/MISC ...................................................................................................................... 238
SIP SETTINGS/SESSION TIMER ................................................................................................... 238
SIP SETTINGS/TCP and TLS ......................................................................................................... 239
SIP SETTINGS/NAT ........................................................................................................................ 240
SIP SETTINGS/TOS........................................................................................................................ 241
STATUS AND REPORTING...................................................................... 243
PBX STATUS ................................................................................................................................... 243
TRUNKS ................................................................................................................................... 243
EXTENSIONS........................................................................................................................... 244
QUEUES .................................................................................................................................. 245
CONFERENCE ROOMS .......................................................................................................... 246
INTERFACES STATUS ............................................................................................................. 247
PARKING LOT .......................................................................................................................... 248
SYSTEM STATUS ........................................................................................................................... 249
GENERAL ................................................................................................................................ 249
NETWORK ............................................................................................................................... 250
STORAGE USAGE ................................................................................................................... 250
RESOURCE USAGE ................................................................................................................ 251
SYSTEM EVENTS .......................................................................................................................... 252
ALERT EVENTS LIST .............................................................................................................. 252
ALERT LOG.............................................................................................................................. 254
ALERT CONTACT .................................................................................................................... 255
CDR................................................................................................................................................. 255
DOWNLOADED CDR FILE ...................................................................................................... 258
STATISTICS ............................................................................................................................. 260
RECORDING FILES ................................................................................................................. 260
API CONFIGURATION ............................................................................................................. 261
UPGRADING AND MAINTENANCE ........................................................ 263
UPGRADING ................................................................................................................................... 263
UPGRADING VIA NETWORK .................................................................................................. 263
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UPGRADING VIA LOCAL UPLOAD ......................................................................................... 264
NO LOCAL FIRMWARE SERVERS ......................................................................................... 266
BACKUP .......................................................................................................................................... 266
LOCAL BACKUP ...................................................................................................................... 266
DATA SYNC .............................................................................................................................. 267
RESTORE CONFIGURATION FROM BACKUP FILE .............................................................. 269
CLEANER ....................................................................................................................................... 270
RESET AND REBOOT .................................................................................................................... 271
SYSLOG .......................................................................................................................................... 271
TROUBLESHOOTING .................................................................................................................... 272
ETHERNET CAPTURE ............................................................................................................ 272
IP PING..................................................................................................................................... 272
TRACEROUTE ......................................................................................................................... 273
ANALOG RECORD TRACE ..................................................................................................... 273
SERVICE CHECK .................................................................................................................... 274
EXPERIENCING THE UCM6100 SERIES IP PBX .................................. 275
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Table of Tables
UCM6100 Series IP PBX User Manual
Table 1: Technical Specifications............................................................................................................... 26
Table 2: UCM6102/UCM6104 Equipment Packaging ............................................................................... 29
Table 3: UCM6108/UCM6116 Equipment Packaging ................................................................................ 29
Table 4: LCD Menu Options ...................................................................................................................... 36
Table 5: UCM6102/UCM6104 LED INDICATORS ..................................................................................... 37
Table 6: UCM6108/UCM6116 LED INDICATORS ..................................................................................... 37
Table 7: User Management - Create New User ........................................................................................ 43
Table 8: Operation Log Column Header .................................................................................................... 45
Table 9: UCM6102 Network Settings->Basic Settings .............................................................................. 46
Table 10: UCM6104 Network Settings->Basic Settings ............................................................................ 48
Table 11: UCM6108/UCM6116 Network Settings->Basic Settings ............................................................ 49
Table 12: UCM6100 Network Settings->802.1X ........................................................................................ 54
Table 13: UCM6100 Network Settings->Static Routes .............................................................................. 54
Table 14: UCM6102 Network Settings->Port Forwarding.......................................................................... 56
Table 15: UCM6100 Firewall->Static Defense->Current Service............................................................... 58
Table 16: Typical Firewall Settings ............................................................................................................ 59
Table 17: Firewall Rule Settings ................................................................................................................ 60
Table 18: UCM6102 Firewall Dynamic Defense ........................................................................................ 61
Table 19: Fail2Ban Settings ...................................................................................................................... 62
Table 20: HTTP Server Settings ................................................................................................................ 70
Table 21: Email Settings............................................................................................................................ 71
Table 22: Time Auto Updating ................................................................................................................... 73
Table 23: Create New Office Time ............................................................................................................ 75
Table 24: Create New Holiday................................................................................................................... 77
Table 25: Auto Provision Settings.............................................................................................................. 86
Table 26: Global Policy Parameters: Localization ..................................................................................... 89
Table 27: Global Policy Parameters: Phone Settings ................................................................................ 90
Table 28: Global Policy Parameters: Contact List ..................................................................................... 90
Table 29: Global Policy Parameters: Maintenance .................................................................................... 92
Table 30: Global Policy Parameters: Network Settings ............................................................................. 94
Table 31: Global Policy Parameters: Customization ................................................................................. 94
Table 32: Create New Template ................................................................................................................ 96
Table 33: Create New Model Template ..................................................................................................... 98
Table 34: SIP Extension Configuration Parameters - Basic Settings........................................................114
Table 35: SIP Extension Configuration Parameters: Media ......................................................................115
Table 36: SIP Extension Configuration Parameters - Features ................................................................116
Table 37: SIP Extension Configuration Parameters - Specific Time .........................................................119
Table 38: IAX Extension Configuration Parameters - Basic Settings .......................................................119
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Table 39: IAX Extension Configuration Parameters - Media ................................................................... 120
Table 40: IAX Extension Configuration Parameters - Features ............................................................... 121
Table 41: IAX Extension Configuration Parameters - Specific Time ........................................................ 123
Table 42: FXS Extension Configuration Parameters - Basic Settings ..................................................... 123
Table 43: FXS Extension Configuration Parameters - Media .................................................................. 124
Table 44: FXS Extension Configuration Parameters - Features .............................................................. 125
Table 45: FXS Extension Configuration Parameters - Specific Time ....................................................... 127
Table 46: Batch Add SIP Extension Parameters ..................................................................................... 128
Table 47: Batch Add IAX Extension Parameters ..................................................................................... 130
Table 48: Analog Trunk Configuration Parameters .................................................................................. 137
Table 49: PSTN Detection for Analog Trunk ............................................................................................ 142
Table 50: Create New SIP Trunk ............................................................................................................. 144
Table 51: SIP Register Trunk Configuration Parameters ......................................................................... 145
Table 52: SIP Peer Trunk Configuration Parameters .............................................................................. 147
Table 53: Create New IAX Trunk ............................................................................................................. 149
Table 54: IAX Register Trunk Configuration Parameters ......................................................................... 150
Table 55: IAX Peer Trunk Configuration Parameters .............................................................................. 151
Table 56: SLA Station Configuration Parameters .................................................................................... 155
Table 57: Outbound Route Configuration Parameters ............................................................................ 159
Table 58: Inbound Rule Configuration Parameters ................................................................................. 162
Table 59: Conference Bridge Configuration Parameters ......................................................................... 165
Table 60: Conference Caller IVR Menu ................................................................................................... 169
Table 61: IVR Configuration Parameters ................................................................................................. 171
Table 62: Voicemail Settings ................................................................................................................... 177
Table 63: Voicemail IVR Menu ................................................................................................................ 178
Table 64: Voicemail Email Settings ......................................................................................................... 179
Table 65: Voicemail Group Settings ........................................................................................................ 181
Table 66: Ring Group Parameters ........................................................................................................... 183
Table 67: Paging/Intercom Group Configuration Parameters .................................................................. 185
Table 68: Call Queue Configuration Parameters ..................................................................................... 187
Table 69: FAX/T.38 Settings .................................................................................................................... 197
Table 70: Follow Me Settings .................................................................................................................. 206
Table 71: Follow Me Options ................................................................................................................... 206
Table 72: DISA Settings .......................................................................................................................... 209
Table 73: Callback Configuration Parameters ..........................................................................................211
Table 74: Event List Settings ................................................................................................................... 213
Table 75: UCM6100 Feature Codes........................................................................................................ 223
Table 76: Internal Options/General ......................................................................................................... 229
Table 77: Internal Options/Jitter Buffer .................................................................................................... 231
Table 78: Internal Options/RTP Settings ................................................................................................. 231
Table 79: Internal Options/Ports Config................................................................................................... 232
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Table 80: Internal Options/STUN Monitor................................................................................................ 234
Table 81: Internal Options/Payload ......................................................................................................... 234
Table 82: IAX Settings/General ............................................................................................................... 235
Table 83: IAX Settings/Registration......................................................................................................... 235
Table 84: IAX Settings/Static Defense ..................................................................................................... 236
Table 85: SIP Settings/General ............................................................................................................... 237
Table 86: SIP Settings/Misc .................................................................................................................... 238
Table 87: SIP Settings/Session Timer ..................................................................................................... 238
Table 88: SIP Settings/TCP and TLS ...................................................................................................... 239
Table 89: SIP Settings/NAT ..................................................................................................................... 240
Table 90: SIP Settings/ToS...................................................................................................................... 241
Table 91: Trunk Status............................................................................................................................. 243
Table 92: Extension Status ...................................................................................................................... 244
Table 93: Agent Status ............................................................................................................................ 246
Table 94: Interface Status Indicators ....................................................................................................... 247
Table 95: Parking Lot Status ................................................................................................................... 248
Table 96: System Status->General.......................................................................................................... 249
Table 97: System Status->Network ......................................................................................................... 250
Table 98: CDR Filter Criteria ................................................................................................................... 256
Table 99: CDR Statistics Filter Criteria .................................................................................................... 260
Table 100: API Configuration Files .......................................................................................................... 261
Table 101: Network Upgrade Configuration ............................................................................................ 263
Table 102: Data Sync Configuration ........................................................................................................ 268
Table 103: Cleaner Configuration............................................................................................................ 270
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Table of Figures
UCM6100 Series IP PBX User Manual
Figure 1: UCM6102/UCM6104/UCM6108/UCM6116 ................................................................................ 25
Figure 2: UCM6102 Front View ................................................................................................................. 30
Figure 3: UCM6102 Back View ................................................................................................................. 30
Figure 4: UCM6104 Front View................................................................................................................. 31
Figure 5: UCM6104 Back View ................................................................................................................. 31
Figure 6: UCM6108 Front View ................................................................................................................. 32
Figure 7: UCM6108 Back View ................................................................................................................. 32
Figure 8: UCM6116 Front View ................................................................................................................. 33
Figure 9: UCM6116 Back View ................................................................................................................. 33
Figure 10: UCM6116 Web GUI Login Page .............................................................................................. 38
Figure 11: UCM6100 Web GUI Language ................................................................................................ 40
Figure 12: User Management Page Display When Logged in with Privilege “Super Admin” .................... 42
Figure 13: User Management Page Display When Logged in with Privilege “Admin” ............................... 42
Figure 14: Create New User ..................................................................................................................... 43
Figure 15: User Management – New Users .............................................................................................. 43
Figure 16: Multiple User Operation Error Prompt ...................................................................................... 44
Figure 17: Operation Logs ........................................................................................................................ 44
Figure 18: Operation Logs Filter ............................................................................................................... 45
Figure 19: UCM6102 Network Interface Method: Route ........................................................................... 50
Figure 20: UCM6102 Network Interface Method: Switch .......................................................................... 51
Figure 21: UCM6102 Network Interface Method: Dual ............................................................................. 52
Figure 22: UCM6100 Using 802.1X as Client ........................................................................................... 53
Figure 23: UCM6100 Using 802.1X EAP-MD5 ......................................................................................... 53
Figure 24: UCM6104 Static Route Sample ............................................................................................... 55
Figure 25: UCM6104 Static Route Configuration ...................................................................................... 56
Figure 26: UCM6102 Port Forwarding Configuration ................................................................................ 57
Figure 27: GXP2160 Web Access Using UCM6102 Port Forwarding ....................................................... 58
Figure 28: Create New Firewall Rule ........................................................................................................ 60
Figure 29: Configure Dynamic Defense .................................................................................................... 62
Figure 30: LDAP Server Configurations .................................................................................................... 64
Figure 31: Default LDAP Phonebook DN .................................................................................................. 65
Figure 32: Default LDAP Phonebook Attributes ........................................................................................ 65
Figure 33: LDAP Server: LDAP Phonebook .............................................................................................. 66
Figure 34: Add LDAP Phonebook ............................................................................................................. 66
Figure 35: Edit LDAP Phonebook ............................................................................................................. 67
Figure 36: Import Phonebook.................................................................................................................... 67
Figure 37: Phonebook CSV File Format ................................................................................................... 68
Figure 38: LDAP Phonebook After Import ................................................................................................. 68
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Figure 39: Export Selected LDAP Phonebook .......................................................................................... 69
Figure 40: GXP2200 LDAP Phonebook Configuration .............................................................................. 70
Figure 41: UCM6100 Email Settings ......................................................................................................... 72
Figure 42: Set Time Manually ................................................................................................................... 74
Figure 43: Create New Office Time ........................................................................................................... 75
Figure 44: Settings->Time Settings->Office Time ...................................................................................... 76
Figure 45: Create New Holiday ................................................................................................................. 77
Figure 46: Settings->Time Settings->Holiday............................................................................................ 78
Figure 47: Settings->Recordings Storage ................................................................................................. 79
Figure 48: Recordings Storage Prompt Information .................................................................................. 80
Figure 49: Recording Storage Category .................................................................................................... 80
Figure 50: Login Timeout Settings ............................................................................................................ 81
Figure 51: Zero Config Configuration Architecture for End Point Device .................................................. 84
Figure 52: UCM6100 Zero Config ............................................................................................................. 85
Figure 53: Auto Provision Settings ............................................................................................................ 86
Figure 54: Auto Discover ........................................................................................................................... 88
Figure 55: Discovered Devices ................................................................................................................. 88
Figure 56: Global Policy Categories .......................................................................................................... 89
Figure 57: Edit Global Template ................................................................................................................ 96
Figure 58: Edit Model Template ................................................................................................................ 99
Figure 59: Template Management .......................................................................................................... 100
Figure 60: Create New Device ................................................................................................................ 101
Figure 61: Manage Devices .................................................................................................................... 101
Figure 62: Edit Device ............................................................................................................................. 102
Figure 63: Edit Customize Device Settings ............................................................................................. 104
Figure 64: Add P Value in Customize Device Settings ............................................................................ 105
Figure 65: Modify Selected Devices - Same Model ................................................................................ 106
Figure 66: Modify Selected Devices - Different Models........................................................................... 107
Figure 67: Device List in Zero Config ...................................................................................................... 108
Figure 68: Zero Config Sample - Global Policy ....................................................................................... 109
Figure 69: Zero Config Sample - Device Preview 1 .................................................................................110
Figure 70: Zero Config Sample - Device Preview 2 ................................................................................. 111
Figure 71: Zero Config Sample - Device Preview 3 .................................................................................112
Figure 72: Create New Device .................................................................................................................113
Figure 73: Export Extensions .................................................................................................................. 133
Figure 74: Import Extensions .................................................................................................................. 134
Figure 75: Email To User - Prompt Information ....................................................................................... 135
Figure 76: Account Registration Information and QR Code .................................................................... 135
Figure 77: LDAP Client Information and QR Code .................................................................................. 136
Figure 78: UCM6100 FXO Tone Settings ................................................................................................ 141
Figure 79: UCM6100 PSTN Detection .................................................................................................... 141
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Figure 80: UCM6100 PSTN Detection: Auto Detect ................................................................................ 142
Figure 81: UCM6100 PSTN Detection: Semi-Auto Detect ...................................................................... 142
Figure 82: DOD extension selection ....................................................................................................... 153
Figure 83: Edit DOD................................................................................................................................ 154
Figure 84: SLA Station ............................................................................................................................ 155
Figure 85: Enable SLA Mode for Analog Trunk ....................................................................................... 157
Figure 86: Analog Trunk with SLA Mode Enabled ................................................................................... 157
Figure 87: SLA Example - SLA Station .................................................................................................... 157
Figure 88: SLA Example - MPK Configuration ........................................................................................ 158
Figure 89: Blacklist Configuration Parameters ........................................................................................ 164
Figure 90: Blacklist csv File..................................................................................................................... 164
Figure 91: Conference Invitation From Web GUI .................................................................................... 167
Figure 92: Conference Recording ........................................................................................................... 170
Figure 93: Click on Prompt to Create IVR Prompt .................................................................................. 173
Figure 94: Record New Custom Prompt ................................................................................................. 173
Figure 95: Upload Custom Prompt .......................................................................................................... 174
Figure 96: Language Settings for Voice Prompt...................................................................................... 175
Figure 97: Voice Prompt Package List .................................................................................................... 176
Figure 98: New Voice Prompt Language Added ..................................................................................... 176
Figure 99: Voicemail Email Settings ........................................................................................................ 180
Figure 100: Voicemail Group................................................................................................................... 181
Figure 101: Ring Group .......................................................................................................................... 183
Figure 102: Ring Group Configuration .................................................................................................... 184
Figure 103: Paging/Intercom Group ........................................................................................................ 185
Figure 104: Page/Intercom Group Settings ............................................................................................. 186
Figure 105: Call Queue ........................................................................................................................... 187
Figure 106: Agent Login Settings ............................................................................................................ 190
Figure 107: Edit Extension Group ........................................................................................................... 191
Figure 108: Select Extension Group in Outbound Route ........................................................................ 192
Figure 109: Edit Pickup Group ................................................................................................................ 193
Figure 110: Edit Pickup Feature Code .................................................................................................... 194
Figure 111: Music On Hold Default Class................................................................................................ 195
Figure 112: Configure Analog Trunk without Fax Detection .................................................................... 198
Figure 113: Configure Extension for Fax Machine .................................................................................. 199
Figure 114: Configure Inbound Rule for Fax ........................................................................................... 199
Figure 115: Create Fax Extension ........................................................................................................... 200
Figure 116: Inbound Route to Fax Extension .......................................................................................... 200
Figure 117: Create Follow Me ................................................................................................................. 205
Figure 118: Edit Follow Me ...................................................................................................................... 205
Figure 119: Configure One-Key Dial ....................................................................................................... 207
Figure 120: One-Key Dial Destinations ................................................................................................... 208
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Figure 121: Create New DISA ................................................................................................................. 209
Figure 122: Create New Event List ......................................................................................................... 214
Figure 123: Create Dial By Name Group ................................................................................................ 215
Figure 124: Dial By Name Group In IVR Key Pressing Events ............................................................... 216
Figure 125: Dial By Name Group In Inbound Rule .................................................................................. 217
Figure 126: Configure Extension First Name and Last Name ................................................................. 218
Figure 127: Status->PBX Status->Active Calls - Ringing ........................................................................ 219
Figure 128: Status->PBX Status->Active Calls – Call Established .......................................................... 219
Figure 129: Configure to Monitor an Active Call ...................................................................................... 220
Figure 130: Download Recording File from CDR Page ........................................................................... 227
Figure 131: Download Recording File from Recording Files Page .......................................................... 227
Figure 132: FXS Ports Signaling Preference .......................................................................................... 232
Figure 133: FXO Ports ACIM Settings..................................................................................................... 232
Figure 134: Status->PBX Status.............................................................................................................. 243
Figure 135: Trunk Status ......................................................................................................................... 243
Figure 136: Extension Status .................................................................................................................. 244
Figure 137: Queue Status ....................................................................................................................... 246
Figure 138: Conference Room Status ..................................................................................................... 247
Figure 139: UCM6116 Interfaces Status ................................................................................................. 247
Figure 140: Parking Lot Status ................................................................................................................ 248
Figure 141: System Status->Storage Usage ........................................................................................... 251
Figure 142: System Status->Resource Usage ........................................................................................ 251
Figure 143: System Events->Alert Events Lists: Disk Usage .................................................................. 252
Figure 144: System Events->Alert Events Lists: Memory Usage ............................................................ 253
Figure 145: System Events->Alert Events Lists: System Reboot ............................................................ 253
Figure 146: System Events->Alert Events Lists: System Crash .............................................................. 253
Figure 147: System Events->Alert Log ................................................................................................... 254
Figure 148: Filter for Alert Log................................................................................................................. 255
Figure 149: CDR Filter ............................................................................................................................ 256
Figure 150: Call Report ........................................................................................................................... 256
Figure 151: Call Report Entry with Audio Recording File ........................................................................ 257
Figure 152: Automatic Download Settings .............................................................................................. 258
Figure 153: Downloaded CDR File Sample - Call To Shows "s" ............................................................. 258
Figure 154: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ............................... 259
Figure 155: Downloaded CDR File Sample - Source Channel and Dest Channel 2 ............................... 259
Figure 156: Downloaded CDR File Sample - Source Channel and Dest Channel 3 ............................... 259
Figure 157: CDR Statistics ...................................................................................................................... 260
Figure 158: CDR->Recording Files ......................................................................................................... 261
Figure 159: Network Upgrade ................................................................................................................. 263
Figure 160: Local Upgrade...................................................................................................................... 264
Figure 161: Upgrading Firmware Files .................................................................................................... 265
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Figure 162: Reboot UCM6100 ................................................................................................................ 265
Figure 163: Local Backup ....................................................................................................................... 267
Figure 164: Data Sync ............................................................................................................................ 268
Figure 165: Restore UCM6100 from Backup File ................................................................................... 269
Figure 166: Cleaner ................................................................................................................................ 270
Figure 167: Reset and Reboot ................................................................................................................ 271
Figure 168: Ethernet Capture.................................................................................................................. 272
Figure 169: PING .................................................................................................................................... 273
Figure 170: Traceroute ............................................................................................................................ 273
Figure 171: Troubleshooting Analog Trunks ............................................................................................ 274
Figure 172: Service Check ...................................................................................................................... 274
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GNU GPL INFORMATION
UCM6100 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_gpl
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CHANGE LOG
This section documents significant changes from previous versions of the UCM6100 user manuals. Only
major new features or major document updates are listed here. Minor updates for corrections or editing are
not documented here.
FIRMWARE VERSION 1.0.8.11

Added recording files store management. [RECORDINGS STORAGE]

Added AMI access (restricted). [ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)]

Added LDAP Phonebook Import/Export support. [LDAP PHONEBOOK]

Added Callback feature. [CALLBACK FEATURE]

Added Busy Camp-on (Call Completion) feature. [BUSY CAMP-ON]

Added external number calling via extension. [Table 36: SIP Extension Configuration Parameters Feature]

Added end point firmware management support to allow end point device to upgrade via
UCM6100 local storage or USB media. [Table 29: Global Policy Parameters: Maintenance]

Added link to access phone web GUI in Zero Config. [MANAGE DEVICES]

Added model update support to download model templates for surveillance and GSwave. [MODEL
UPDATE]

Added filter for searching in Alert Log. [ALERT LOG]

Added DAHDI Out Line Selection. [Table 48: Analog Trunk Configuration Parameters]

Added automatic CDR download. [CDR]

Added multi-user management and concurrent user login support. [USER MANAGEMENT]

Added operation log. [OPERATION LOG]
Added more options on Global Policy settings. [

Table 27: Global Policy Parameters: Phone Settings]

Added option to specify web access session timer. [LOGIN TIMEOUT SETTINGS]

Added call monitor/barge-in support. [ACTIVE CALLS AND MONITOR]

Added GSwave QR code support. [EMAIL TO USER]

Added ability to add customized Pvalue on Model Template. [MODEL TEMPLATES]

Updated maximum length of web UI login password to 30 characters. [CHANGE PASSWORD]

Added Fax gateway support for Fax mode on FXS and FXO port. [Table 43: FXS Extension
Configuration Parameters - Media] [Table 48: Analog Trunk Configuration Parameters]

Added “NAT” option and “Send PPI Header” option in SIP register trunk configuration. [Table 51:
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SIP Register Trunk Configuration Parameters]

Added REC API (access for call recording files). [API CONFIGURATION]

Added “Synchronize All Data” option in Data Sync. [DATA SYNC]

Added “Delect Selected Sound Files” option for Music On Hold files. [MUSIC ON HOLD]
FIRMWARE VERSION 1.0.7.11

Added office time and holiday setting. [OFFICE TIME] [HOLIDAY]

Added new architecture for auto provisioning via Zero Config. [PROVISIONING]

Added Alert-info configuration for distinctive ringing on inbound route. [Table 58: Inbound Rule
Configuration Parameters]

Added SLA support to monitor FXO trunk. [SLA STATION]

Added Follow Me support. [FOLLOW ME]

Added One-Key Dial support. [ONE-KEY DIAL]

Added FXO Dial Delay option for analog trunk. [Table 48: Analog Trunk Configuration Parameters]

Added Music On Hold selection per extension. [EXTENSIONS]

Added Call Forward Time Condition per extension. [EXTENSIONS]

Added custom prompt support for ring group announcement. [Table 66: Ring Group Parameters]

Added custom prompt support for paging/intercom group announcement. [Table 67:
Paging/Intercom Group Configuration Parameters]

Added options to delete voicemail and Fax files via cleaner. [CLEANER]

Added DTMF configuration per trunk. [VOIP TRUNKS]

Added call queue Music On Hold customization. [Table 68: Call Queue Configuration Parameters]

Added call queue destination support. [Table 68: Call Queue Configuration Parameters]

Added batch add blacklist file support. [BLACKLIST CONFIGURATIONS]
FIRMWARE VERSION 1.0.6.11

Added service check options under web UI->Maintenance->Troubleshooting->Service Check
page to regularly check Asterisk service.
FIRMWARE VERSION 1.0.6.10

Added static routes function. [STATIC ROUTES]

Added option to provision end devices’ date format, time format and time zone in zero config.

Added option to disable extension/trunk.

Added TEL URL support for extension/trunk.

Added option to dial trunk password per extensions.
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
Added export extension and import extension function. [EXPORT EXTENSIONS] [IMPORT
EXTENSIONS]

Added option "Need Registration" for SIP register trunk.

Added option "The Maximum Number of Call Line" for trunk.

Added Dial By Name. [DIAL BY NAME]

Added voicemail password and Email address for voicemail group extension.

Added auto record support for ring group and call queue.

Added VFax file display, download and delete interface in web UI.

Changed web page name from "Hardware Config" to "Ports Config".

Added payload configuration for audio/video codecs. [INTERNAL OPTIONS/PAYLOAD]

Added activity calls status on web UI status page.

Added CDR API support. [API CONFIGURATION]

Added more alert events support such as Register SIP Failed, Register SIP Trunk Failed, Restore
Config, User Login Success, User Login Failed, SIP Internal Call Failure and etc. [ALERT
EVENTS LIST]
FIRMWARE VERSION 1.0.5.19

Added built-in data migration tool to support upgrading from 1.0.4.7 to 1.0.5.19 without factory
reset.

Added "Direct Dial Voicemail Prefix" feature code back. [Table 75: UCM6100 Feature Codes]

Changed valid range for option "Current Disconnect Threshold". [Table 48: Analog Trunk
Configuration Parameters]
FIRMWARE VERSION 1.0.5.14

New backend data structure and web UI performance improvement. 1.0.5.14 is not compatible
with previous firmware versions. Once upgraded to 1.0.5.14, the device needs to be FULLY
RESET and RE-CONFIGURED MANUALLY.

Added traditional Chinese language for web UI. [WEB GUI LANGUAGES]

Updated LDAP configuration example. [LDAP SERVER]

Added "Enable Filter Source Caller ID" and "Custom Dynamic Route" options for outbound route
settings. [Table 57: Outbound Route Configuration Parameters]

Added more language support for voice prompt. [LANGUAGE SETTINGS FOR VOICE PROMPT]

Added "Ring Group Destination" for ring group configuration. [Table 66: Ring Group Parameters]

Added "Extension Groups" section in web UI. [EXTENSION GROUPS]

Added "Pickup Groups" section in web UI. [PICKUP GROUPS]

Added BLF function description. [BLF AND EVENT LIST]

Updated default extension range. [Table 76: Internal Options/General]
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
Added sample descriptions for downloaded CDR file. [DOWNLOADED CDR FILE]
FIRMWARE VERSION 1.0.4.7

Asterisk updated to version 1.8.23.1.

Added DID routing support for incoming calls. [Table 50: Create New SIP Trunk]

Added DOD routing support. [Direct Outward Dialing (DOD)]

Added GXP one-button Voicemail access. [Table 34: SIP Extension Configuration Parameters]

Added option "Skip voicemail password verification" on extension edit page. [Table 34: SIP
Extension Configuration Parameters]

Added Hot-Desking Support. [Table 34: SIP Extension Configuration Parameters]

Added one-button on-demand call recording for GXP

Add new option to enable or disable "FXS TISS Override" on Hardware Config page. [Table 79:
Internal Options/Ports Config]

Added more modes for FXS Two-Wire Impedance Synthesis

Added LDAP Sync manual trigger function and synced date displaying. [VOIP TRUNKS]

Improved LDAP Sync function, added retrying, file verifying and progress displaying function

Added option "Pick Extension Period" on auto-provision settings page of Zero Config

Added multiple extension assignment support on device edit page of Zero Config

Added "Reset All Extensions" button at the Zero Config page to recycle all assigned extensions.
[AUTO PROVISIONING]

Added system crash alarm, core dump detection and allow users to download core dump file.
[Figure 146: System Events->Alert Events Lists: System Crash]

Add "Keep Trunk CID" option for VoIP trunks, and keep the priorities: DOD -> Extension CallerID
-> Trunk CallerID -> Global CallerID. [Table 50: Create New SIP Trunk]
FIRMWARE VERSION 1.0.3.13

Added Fail2Ban support for SIP authentication. [FAIL2BAN]

Added voice prompt "Language" selection and "Auto Record" option for extension. [Table 34: SIP
Extension Configuration Parameters] [Table 46: Batch Add SIP Extension Parameters]

Added "Auto Record" option for trunk.

Added "Digit Timeout" option and voice prompt "Language" selection for IVR. [Table 61: IVR
Configuration Parameters]

Added "Direct Dial Voicemail Prefix" feature code to directly dial or transfer to extension's
voicemail. [Table 75: UCM6100 Feature Codes]

Added "Enforce Strong Passwords" option. [Table 76: Internal Options/General]

Added FXS MWI Mode. [Table 79: Internal Options/Ports Config]

Added system events with alert and Email notification support. [SYSTEM EVENTS]
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
Added new web page for recording files. [RECORDING FILES]
FIRMWARE VERSION 1.0.2.21

Added weight information. [Table 1: Technical Specifications]

Added NTP server support. [NTP SERVER]

Added Czech language for web GUI display. [WEB GUI LANGUAGES]

Added VLAN support. "Layer 2 QoS 802.1Q/VLAN tag" and "Layer 2 QoS 802.1p priority value"
options for network port settings are added. [NETWORK SETTINGS]

Updated LDAP client configurations information.

Added sample Email settings. [EMAIL SETTINGS]

Added manual time settings. [TIME SETTINGS]

Added "Enable Pick Extension" and "Extension Segment" options for auto provisioning settings.

Changed one of the discovery method from "SIP MESSAGE (OPTIONS)" to "SIP MESSAGE
(NOTIFY)" in zero-config feature.

Added pickup group feature. [Table 75: UCM6100 Feature Codes]

Added PSTN detection instructions for "Auto Detect" and "Semi-auto Detect". [PSTN
DETECTION]

Added "Auth ID" option for SIP register trunk configuration. [Table 50: Create New SIP Trunk]

Added LDAP sync options for peer SIP trunk. [Table 50: Create New SIP Trunk]

Changed the default setting of outbound route "Privilege Level" from "Internal" to "International" to
avoid potential misconfiguration and security risk.

Added DISA and Fax to IVR key press event options. [Table 61: IVR Configuration Parameters]

Added "Min Message Time" option in voicemail settings. [Table 62: Voicemail Settings]

Added Fax setting samples. [FAX/T.38]

Added DISA support for inbound route and IVR.

Added Event List support to monitor local extensions and remote extensions. [BLF AND EVENT
LIST]

Added feature code *0 for "Disconnect". [Table 75: UCM6100 Feature Codes]

Added feature code *8 for "Pickup Extension" in pickup group feature. [Table 75: UCM6100
Feature Codes]

Added "Record Prompt" and "Custom Name of Pickup Group" options in internal options. [Table
76: Internal Options/General]

Added warning information for "Allow Guest Call" option to avoid potential security risk caused by
misconfiguration. [Table 82: IAX Settings/General]

Changed reset mode to two mode "User Data" and "All". [RESET AND REBOOT]
FIRMWARE VERSION 1.0.1.22
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
This is the initial version.
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WELCOME
Thank you for purchasing Grandstream UCM6100 series IP PBX appliance. The UCM6100 series IP PBX
is an innovative IP PBX appliance designed for small to medium business. Powered by an advanced
hardware platform with robust system resources, the UCM6100 offers a highly versatile state-of-the-art
Unified Communication (UC) solution for converged voice, video, data, fax and video surveillance
application needs. Incorporating industry-leading features and performance, the UCM6100 offers quick
setup, deployment with ease and unrivaled reliability all at an unprecedented price point.
Caution:
Changes or modifications to this product not expressly approved by Grandstream, or operation of this
product in any way other than as detailed by this User Manual, could void your manufacturer warranty.
Warning:
Please do not use a different power adaptor with the UCM6100 as it may cause damage to the products
and void the manufacturer warranty.
This document is subject to change without notice. The latest electronic version of this user manual is
available for download here:
http://www.grandstream.com/support
Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for
any purpose without the express written permission of Grandstream Networks, Inc. is not permitted.
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PRODUCT OVERVIEW
FEATURE HIGHTLIGHTS

1GHz ARM Cortex A8 application processor, large memory (512MB DDR RAM, 4GB NAND Flash),
and dedicated high performance multi-core DSP array for advanced voice processing.

Integrated 2/4/8/16 PSTN trunk FXO ports, 2 analog telephone FXS ports with lifeline capability in
case of power outage, and up to 50 SIP trunk options.

Gigabit network port(s) with integrated PoE, USB, SD; integrated NAT router with advanced QoS
support (UCM6102 only).

Supports a wide range of popular voice codes (including G.711 A-law/U-law, G.722, G.723.1, G.726,
G.729A/B, iLBC, GSM), video codec (including H.264, H.263, H.263+), and Fax (T.38).

Hardware DSP based 128ms-tail-length carrier-grade line echo cancellation (LEC).

Supports up to 500 SIP endpoint registration, up to 60 concurrent calls and up to 32 conference
attendees.

Flexible dial plan, call routing, site peering, call recording.

Automated detection and provisioning of IP phones, video phones, ATA and other endpoints for easy
deployment.

Hardware encryption accelerator to ensure strongest security protection using SRTP, TLS, and
HTTPS.
Figure 1: UCM6102/UCM6104/UCM6108/UCM6116
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TECHNICAL SPECIFICATIONS
Table 1: Technical Specifications
Interfaces
Analog Telephone FXS Ports
PSTN Line FXO Ports
Network Interfaces
2 ports (both with lifetime capability in case of power outage)

UCM6102: 2 ports

UCM6104: 4 ports

UCM6108: 8 ports

UCM6116: 16 ports

UCM6102/6104: Dual 10M/100M/1000M RJ45 Ethernet ports with
integrated PoE Plug (IEEE 802.3at-2009)

UCM6108/6116: Single 10M/100M/1000M RJ45 Ethernet port with
integrated PoE Plug (IEEE 802.3at-2009)
NAT Router
Yes, UCM6102 only
Peripheral Ports
USB, SD/SDHC (VFAT)
LED Indicators
Power/Ready, Network, PSTN Line, USB, SD
LCD Display
128x32 graphic LCD with DOWN and OK button
Reset Switch
Yes
Voice/Video Capabilities
Voice-over-Packet
Capabilities
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; T.38
Video Codecs
H.264, H.263, H.263+
QoS
Layer 3 QoS
Signaling and Control
DTMF Methods
Provisioning Protocol and
Plug-and-Play
Network Protocols
Disconnect Methods
Firmware Version 1.0.8.12
In Audio, RFC2833, and SIP INFO
TFTP/HTTP/HTTPS,
auto-discovery
and
auto-provisioning
of
Grandstream IP endpoints via ZeroConfig (DHCP Option 66/multicast SIP
SUBSCRIBE/mDNS)
TCP/UDP/IP, RTP/RTCP, ICMP, ARP, DNS, DDNS, DHCP, NTP, TFTP,
SSH, HTTP/HTTPS, PPPoE, SIP (RFC3261), STUN, SRTP, TLS
Call Progress Tone, Polarity Reversal, Hook Flash Timing, Loop Current
Disconnect, Busy Tone
UCM6100 Series IP PBX User Manual
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Security
Media
SRTP, TLS, HTTPS, SSH
Physical
Universal Power Supply
Environmental
Dimensions
Weight
Mounting

Output: 12VDC, 1.5A

Input: 100-240VAC, 50-60Hz

Operating: 32 - 104oF / 0 - 40oC, 10-90% (non-condensing)

Storage: 14 - 140oF / -10 - 60oC

UCM6102/6104: 226mm (L) x 155mm (W) x 34.5mm (H)

UCM6108/6116: 440mm (L) x 185mm (W) x 44mm (H)

UCM6102: Unit weight 0.51kg, Package weight 0.94kg

UCM6104: Unit weight 0.51kg, Package weight 0.94kg

UCM6108: Unit weight 2.23kg, Package weight 3.09kg

UCM6116: Unit weight 2.27kg, Package weight 3.14kg

UCM6102/6104: Wall mount and Desktop

UCM6108/6116: Rack mount and Desktop
Additional Features
Multi-language Support
Yes, English/Chinese/Spanish/French/German/Russian/Italian for Web
GUI; Customizable IVR to support any language
Caller ID
Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 - BT, NTT Japan
Polarity Reversal/ Wink
Yes, with enable/disable option upon call establishment and termination
Call Center
Customizable Auto Attendant
Concurrent Calls
Conference Bridges
Multiple configurable call queues, automatic call distribution (ACD)
based on agent skills/availability busy level, in-queue announcement
Up to 5 layers of IVR (Interactive Voice Response)

UCM6102: Up to 30 simultaneous calls

UCM6104: Up to 45 simultaneous calls

UCM6108/6116: Up to 60 simultaneous calls

UCM6102/6104: Up to 3 password-protected conference bridges
allowing up to 25 simultaneous PSTN or IP participants

UCM6108/6116: Up to 6 password-protected conference bridges
allowing up to 32 simultaneous PSTN or IP participants
Call Features
Compliance
Call
call
forward,
call
transfer,
DND,
ring/hunt
group,

FCC: Part 15 (CFR 47) Class B, Part 68

CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3,
EN60950-1, TBR21, RoHS

Firmware Version 1.0.8.12
park,
paging/intercom and etc
A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS
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60950, AS/ACIF S002 and ITU-T K.21 (Basic Level)

Firmware Version 1.0.8.12
UL 60950 (power adapter)
UCM6100 Series IP PBX User Manual
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INSTALLATION
Before deploying and configuring the UCM6100 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 UCM6100 series.
EQUIPMENT PACKAGING
Table 2: UCM6102/UCM6104 Equipment Packaging
Main Case
Yes (1)
Power Adaptor
Yes (1)
Ethernet Cable
Yes (1)
Quick Installation Guide
Yes (1)
GPL License
Yes (1)
Table 3: UCM6108/UCM6116 Equipment Packaging
Main Case
Yes (1)
Power Adaptor
Yes (1)
Ethernet Cable
Yes (1)
Wall Mount
Yes (2)
Screws
Yes (6)
Quick Installation Guide
Yes (1)
GPL License
Yes (1)
CONNECT YOUR UCM6100
CONNECT THE UCM6102
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Figure 2: UCM6102 Front View
Figure 3: UCM6102 Back View
To set up the UCM6102, follow the steps below:
1. Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6102.
2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6102. Insert
the main plug of the power adapter into a surge-protected power outlet.
4. Wait for the UCM6102 to boot up. The LCD in the front will show the device hardware information
when the boot process is done.
5. Once the UCM6102 is successfully connected to network, the LED indicator for WAN in the front will
be in solid green and the LCD shows up the IP address.
6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and
Fax) to the FXS ports.
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CONNECT THE UCM6104
Figure 4: UCM6104 Front View
Figure 5: UCM6104 Back View
To set up the UCM6104, follow the steps below:
1. Connect one end of an RJ-45 Ethernet cable into the LAN 1 port of the UCM6104.
2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6104. Insert
the main plug of the power adapter into a surge-protected power outlet.
4. Wait for the UCM6104 to boot up. The LCD in the front will show the device hardware information
when the boot process is done.
5. Once the UCM6104 is successfully connected to network, the LED indicator for LAN 1 in the front will
be in solid green and the LCD shows up the IP address.
6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and
Fax) to the FXS ports.
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CONNECT THE UCM6108
To set up the UCM6108, follow the steps below:
1. Connect one end of an RJ-45 Ethernet cable into the LAN port of the UCM6108.
2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6108. Insert
the main plug of the power adapter into a surge-protected power outlet.
4. Wait for the UCM6108 to boot up. The LCD in the front will show the device hardware information
when the boot process is done.
5. Once the UCM6108 is successfully connected to network, the LED indicator for NETWORK in the front
will be in solid green and the LCD shows up the IP address.
6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and
Fax) to the FXS ports.
Figure 6: UCM6108 Front View
Figure 7: UCM6108 Back View
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CONNECT THE UCM6116
Figure 8: UCM6116 Front View
Figure 9: UCM6116 Back View
To set up the UCM6116, follow the steps below:
1. Connect one end of an RJ-45 Ethernet cable into the LAN port of the UCM6116.
2. Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.
3. Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6116. Insert
the main plug of the power adapter into a surge-protected power outlet.
4. Wait for the UCM6116 to boot up. The LCD in the front will show the device hardware information
when the boot process is done.
5. Once the UCM6116 is successfully connected to network, the LED indicator for NETWORK in the front
will be in solid green and the LCD shows up the IP address.
6. (Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and
Fax) to the FXS ports.
SAFETY COMPLIANCES
The UCM6100 series IP PBX complies with FCC/CE and various safety standards. The UCM6100 power
adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM6100
package only. The manufacturer’s warranty does not cover damages to the device caused by unsupported
power adapters.
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WARRANTY
If the UCM6100 series IP PBX was purchased from a reseller, please contact the company where the
device was purchased for replacement, repair or refund. If the device was purchased directly from
Grandstream, contact our Technical Support Team for a RMA (Return Materials Authorization) number
before the product is returned. Grandstream reserves the right to remedy warranty policy without prior
notification.
Warning:
Use the power adapter provided with the UCM6100 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 UCM6100 series provides LCD interface, LED indication and web GUI configuration interface.

The LCD displays hardware, software and network information. Users could also navigate in the LCD
menu for device information and basic network configuration.

The LED indication at the front of the device provides interface connection and activity status.

The web GUI gives users access to all the configurations and options for UCM6100 series setup.
This section provides step-by-step instructions on how to use the LCD menu, LED indicators and Web GUI
of the UCM6100 series. Once the basic settings are done, users could start making calls from UCM6100
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., UCM6116),
hardware version (e.g., V1.5A) and IP address. Press "Down" button and the system time will be
displayed as well.

Menu Access
Press "OK" button to start browsing menu options. Please see menu options in [Table 4: LCD Menu
Options].

Menu Navigation
Press the "Down" arrow key to browser different menu options. Press the "OK" button to select an
entry.

Exit
If "Back" option is available in the menu, select it to go back to the previous menu. For "Device Info"
"Network Info" and "Web Info" which do not have "Back" option, simply press the "OK" button to go
back to the previous menu. Also, the LCD will display default idle screen after staying in menu option
for 15 seconds.

LCD Backlight
The LCD backlight will be on upon key pressing. The backlight will go off after the LCD stays in idle for
30 seconds.
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Table 4: LCD Menu Options
View Events
Device Info

Critical Events

Other Events

Hardware: Hardware version number

Software: Software version number

P/N: Part number

WAN MAC: WAN side MAC address (UCM6102 only)

LAN MAC: LAN side MAC address

Uptime: System up time
For UCM6104/UCM6108/UCM6116:

LAN Mode: DHCP, Static IP, or PPPoE

LAN IP: IP address

LAN Subnet Mask
For UCM6102:
Network Info

WAN Mode: DHCP, Static IP, or PPPoE

WAN IP: IP address

WAN Subnet Mask

LAN IP: IP address

LAN Subnet Mask
For UCM6104/UCM6108/UCM6116:
Network Menu

LAN Mode: Select LAN mode as DHCP, Static IP or PPPoE

Static Route Reset: Click to reset the static route setting
For UCM6102:

WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE

Static Route Reset: Click to reset the static route setting

Reboot

Factory Reset

LCD Test Patterns
Press "OK" to start. Then press "Down" button to test different LCD
Factory Menu
patterns. When done, press "OK" button to exit.

Fan Mode
Select "Auto" or "On".

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Select "All On" "All Off" or "Blinking" and check LED status.

RTC Test Patterns
Select "2022-02-22 22:22" or "2011-01-11 11:11" to start the RTC
(Real-Time Clock) test pattern. Then check the system time from LCD idle
screen by pressing "DOWN" button, or from web GUI->System
Status->General page. Reboot the device manually after the RTC test is
done.

Hardware Testing
Select "Test SVIP" to perform SVIP test on the device. This is mainly for
factory testing purpose which verifies the hardware connection inside the
device. The diagnostic result will display in the LCD after the test is done.
Web Info

Protocol: Web access protocol. HTTP or HTTPS. By default it's HTTPS

Port: Web access port number. By default it's 8089
USE THE LED INDICATORS
The UCM6100 has LED indicators in the front to display connection status. The following table shows the
status definitions.
Table 5: UCM6102/UCM6104 LED INDICATORS
LED Indicator
LED Status
LAN
WAN
Solid: Connected
USB
Flashing: Data Transferring
SD
OFF: Not Connected
FXS (Phone/Fax)
FXO (Telco Line)
Table 6: UCM6108/UCM6116 LED INDICATORS
LED
NETWORK
LED Status
Solid: Connected
OFF: Not Connected
ACT
USB
Solid: Connected
SD
Flashing: Data Transferring
Phone (FXS)
OFF: Not Connected
Line (FXO)
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USE THE WEB GUI
ACCESS WEB GUI
The UCM6100 embedded Web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow users to configure the device through a Web browser such as Microsoft IE, Mozilla Firefox,
Google Chrome and etc.
Figure 10: UCM6116 Web GUI Login Page
To access the Web GUI:
1. Connect the computer to the same network as the UCM6100.
2. Ensure the device is properly powered up and shows its IP address on the LCD.
3. Open a Web browser on the computer and enter the web GUI URL in the following format:
http(s)://IP-Address:Port
where the IP-Address is the IP address displayed on the UCM6100 LCD.
By default, the protocol is HTTPS and the Port number is 8089.
For example, if the LCD shows 192.168.40.167, please enter the following in your web browser:
https://192.168.40.167:8089
4. Enter the administrator’s login and password to access the Web Configuration Menu. The default
administrator's username and password is "admin" and "admin". It is highly recommended to change
the default password after login for the first time.
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Note:
By default, the UCM6100 has "Redirect From Port 80" enabled. Therefore, if users type in the UCM6100 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 UCM6100 web GUI->Settings->HTTP
Server.
WEB GUI CONFIGURATIONS
There are four main sections in the Web GUI for users to view the PBX status, configure and manage the
PBX.

Status: Displays PBX status, System Status, System Events and CDR.

PBX: To configure extensions, trunks, call routes, zero config for auto provisioning, call features,
internal options, IAX settings and SIP settings.

Settings: To configure user management, network settings, firewall settings, change password,
LDAP Server, HTTP Server, Email Settings, Time Settings, NTP server, recording storage and login
timeout.

Maintenance: To perform firmware upgrade, backup configurations, cleaner setup, reset/reboot,
syslog setup and troubleshooting.
WEB GUI LANGUAGES
Currently the UCM6100 series web GUI supports 11 languages: English, Simplified Chinese,
Traditional Chinese, Spanish, French, Portuguese, Russian, Italian, Polish, German, Czech.
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 11: UCM6100 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.
MAKE YOUR FIRST CALL
Power up the UCM6100 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 UCM6100 web GUI, go to PBX->Basic/Call Routes->Extensions.
2. Click on "Create New SIP Extension" to create a new extension. You will need User ID, Password and
Voicemail Password information to register and use the extension later.
3. Register the extension on your phone with the SIP User ID, SIP server and SIP Password information.
The SIP server address is the UCM6100 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 UCM6100. System settings are
under “Settings” tag on UCM6100 web GUI. System settings include User Management, Network Settings,
Firewall, Change Password, LDAP server, HTTP server, Email settings, Time Settings, NTP Server,
Recordings Storage and Login Timeout settings.
USER MANAGEMENT
Starting from firmware 1.0.8.11, users could create multiple accounts for different administrators to log in
the UCM6100 web GUI. This can be configured on web GUI->Settings->User Management page.
USER PRIVILEGES
Two privilege levels are supported:

Super Admin
-
This is the highest privilege. Super Admin can access all pages on UCM6100 web GUI, change
configuration for all options and execute all the operations.
-
Super Admin can create, edit and delete one or more new users with “Admin” privilege, as well as
view operation logs generated by other 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.
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Figure 12: User Management Page Display When Logged in with Privilege “Super Admin”

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.
Figure 13: User Management Page Display When Logged in with Privilege “Admin”
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 14: Create New User
Table 7: 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 role of the web UI user. Currently only “Admin” is supported when Super
Admin creates a new user.
Department
Fax
Email Address
First Name
Enter the necessary information to keep a record for this user.
Last Name
Home Number
Phone Number
Once created, the Super Admin can edit the users by clicking on
or delete the user by clicking on
.
Figure 15: User Management – New Users
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CONCURRENT MULTI-USER LOGIN
When there are multiple web UI users created, concurrent multi-user login is supported on the UCM6100.
Multiple users could edit options and have configurations take effect simultaneously. However, if different
users are editing the same option or making the same operation (by clicking on “Apply Changes”), a
prompt will pop up as shown in the following figure.
Figure 16: Multiple User Operation Error Prompt
OPERATION LOG
The Super Admin has the authority to view operation logs on UCM6100 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”.
Figure 17: Operation Logs
The operation log can be sorted and filtered for easy access. Click on the header of each column to sort.
For example, clicking on "Date" will sort the logs according to operation date and time. Clicking on "Date"
again will reverse the order.
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Table 8: 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
.
Figure 18: Operation Logs Filter
The above figure shows an example that operations made by user “support” on device with IP
192.168.40.173 from 2014-11-01 00:00 to 2014-11-06 15:38 are filtered out and displayed.
To delete operation logs, users can perform filtering first and then click on
delete the filtered result of operation logs. Or users can click on
to
to delete all
operation logs at once.
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NETWORK SETTINGS
After successfully connecting the UCM6100 to the network for the first time, users could login the Web GUI
and go to Settings->Network Settings to configure the network parameters for the device.
The network setting options are similar for UCM6108 and UCM6116. Additional network functions and
settings are available for UCM6102 and UCM6104:

UCM6102 supports Route/Switch/Dual mode functions.

UCM6104 supports Switch/Dual mode functions.
In this section, all the available network setting options are listed for each model. Select each tab in web
GUI->Settings->Network Settings page to configure LAN settings, WAN settings (UCM6102 only),
802.1X and Port Forwarding (UCM6102 only).
BASIC SETTINGS
Please refer to the following tables for basic network configuration parameters on UCM6102, UCM6104,
and UCM6108/UCM6116 respectively.
Table 9: UCM6102 Network Settings->Basic Settings
Select "Route", "Switch" or "Dual" mode on the network interface of UCM6102.
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.
Preferred DNS Server
Enter the preferred DNS server address.
WAN (when "Method" is set to "Route")
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Enter the subnet mask address for static IP settings. The default setting is
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255.255.0.0.
Gateway IP
DNS Server 1
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
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. The default value
802.1Q/VLAN Tag
is 0.
Layer 2 QoS 802.1p
Assign the priority value of the layer 2 QoS packets for WAN port. The default
Priority Value
value is 0.
LAN (when Method is set to "Route")
IP Address
Enter the IP address assigned to LAN port. The default setting is 192.168.2.1.
Subnet Mask
Enter the subnet mask. The default setting is 255.255.255.0.
DHCP Server Enable
Enable or disable DHCP server capability. The default setting is "Yes".
DNS Server 1
Enter DNS server address 1. The default setting is 8.8.8.8.
DNS Server 2
Enter DNS server address 2. The default setting is 208.67.222.222.
Allow IP Address From
Enter the DHCP IP Pool starting address. The default setting is 192.168.2.100.
Allow IP Address To
Enter the DHCP IP Pool ending address. The default setting is 192.168.2.254.
Default IP Lease Time
Enter the IP lease time (in seconds). The default setting is 43200.
LAN (when Method is set to "Switch")
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Gateway IP
DNS Server 1
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. The default setting is 0.0.0.0.
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
Priority Value
value is 0.
LAN 1 / LAN 2 (when Method is set to "Dual")
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If "Dual" is selected as "Method", users will need assign the default interface to be
Default Interface
LAN 1 (mapped to UCM6102 WAN port) or LAN 2 (mapped to UCM6102 LAN
port) and then configure network settings for LAN 1/LAN 2. The default interface is
LAN 2.
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
Subnet Mask
Gateway IP
DNS Server 1
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.
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
Priority Value
value is 0.
Table 10: UCM6104 Network Settings->Basic Settings
Select "Switch" or "Dual" mode on the network interface of UCM6104. The default
setting is "Switch".

Switch
LAN 1 port interface will be used for uplink connection. LAN 2 port interface
Method
will be used as bridge for PC connection.

Dual
Both ports can be used for uplink connection. Users will need assign the
default interface in option "Default Interface". Users will need assign LAN 1 or
LAN 2 as the default interface in option "Default Interface" and configure
"Gateway IP" for this interface.
Preferred DNS Server
Enter the preferred DNS server address.
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
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Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
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Gateway IP
DNS Server 1
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
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
Priority Value
value is 0.
LAN 1 / LAN 2 (when Method is set to "Dual")
Default Interface
If "Dual" is selected as "Method", users will need assign the default interface to be
LAN 1 or 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
DNS Server 1
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. The default setting is 0.0.0.0.
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
Priority Value
value is 0.
Table 11: UCM6108/UCM6116 Network Settings->Basic Settings
Preferred DNS Server
Enter the preferred DNS server address.
IP Method
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
IP Address
Enter the IP address for static IP settings.
Subnet Mask
Enter the subnet mask address for static IP settings.
Gateway IP
Enter the gateway IP address for static IP settings.
DNS Server 1
Enter the DNS server 1 address for static IP settings.
DNS Server 2
Enter the DNS server 2 address for static IP settings.
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User Name
Enter the user name to connect via PPPoE.
Password
Enter the password to connect via PPPoE.
Layer 2 QoS
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
Priority Value
value is 0.

Method: Route
When the UCM6102 has method set to Route in network settings, WAN port interface is used for
uplink connection and LAN port interface is used as a router. Please see a sample diagram below.
Figure 19: UCM6102 Network Interface Method: Route

Method: Switch
For UCM6102, WAN port interface is used for uplink connection; LAN port interface is used as bridge
for PC connection. For UCM6104, LAN 1 port interface is used for connection; LAN 2 port interface is
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used as bridge for PC connection.
Figure 20: UCM6102 Network Interface Method: Switch

Method: Dual
For UCM6102, both WAN port and LAN port are used for uplink connection. For UCM6104, 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: UCM6102 Network Interface Method: Dual
802.1X
IEEE 802.1X is an IEEE standard for port-based network access control. It provides an authentication
mechanism to device before the device is allowed to access Internet or other LAN resources. The
UCM6100 supports 802.1X as a supplicant/client to be authenticated. The following diagram and figure
show UCM6100 uses 802.1X mode “EAP-MD5” on WAN port as client in the network to access Internet.
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Figure 22: UCM6100 Using 802.1X as Client
Figure 23: UCM6100 Using 802.1X EAP-MD5
The following table shows the configuration parameters for 802.1X on UCM6100. 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.
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Table 12: UCM6100 Network Settings->802.1X
Select 802.1X mode. The default setting is "Disable". The supported 802.1X
mode are:
802.1X Mode

EAP-MD5

EAP-TLS

EAP-PEAPv0/MSCHAPv2
Identity
Enter 802.1X mode identity information.
MD5 Password
Enter 802.1X mode MD5 password information.
802.1X Certificate
Select 802.1X certificate from local PC and then upload.
802.1X Client
Select 802.1X client certificate from local PC and then upload.
Certificate
STATIC ROUTES
The UCM6100 provides users static routing capability that allows the device to use manually configured
routes, rather than information only from dynamic routing or gateway configured in the UCM6100 web
GUI->Network Settings->Basic Settings to forward traffic. It can be used to define a route when no other
routes are available or necessary, or used in complementary with existing routing on the UCM6100 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
to edit the static route.
to delete the static route.
Table 13: UCM6100 Network Settings->Static Routes
Configure the destination IP address or the destination IP subnet for the
UCM6100 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
Netmask
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Example:
255.255.255.0
Configure the gateway address so that the UCM6100 can reach the destination
via this gateway. Gateway address is optional.
Gateway
Example:
192.168.40.5
Specify the network interface on the UCM6100 to reach the destination using the
static route.
Interface
For UCM6102, LAN interface is eth0; WAN interface is eth1.
For UCM6104, LAN1 interface is eth0; WAN interface is eth1.
For UCM6108/UCM6116, only LAN interface is available.
Static routes configuration can be reset from LCD menu->Network Menu.
The following diagram shows a sample application of static route usage on UCM6104.
Figure 24: UCM6104 Static Route Sample
The network topology of the above diagram is as below:

Network 192.168.69.0 has IP phones registered to UCM6104 LAN 1 address

Network 192.168.40.0 has IP phones registered to UCM6104 LAN 2 address
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
Network 192.168.66.0 has IP phones registered to UCM6104 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 UCM6104. Therefore, we need configure a
static route on the UCM6104 so that the phones in isolated networks can make calls between each other.
Figure 25: UCM6104 Static Route Configuration
PORT FORWORDING (UCM6102 ONLY)
The UCM6102 network interface supports router functions which provides users the ability to do port
forwarding. If the UCM6102 LAN mode is set to "Route" under web GUI->Settings->Network
Settings->Basic Settings page, port forwarding is available for configuration.
The port forwarding configuration is under web GUI->Settings->Network Settings->Port Forwarding
page. Please see related settings in the table below.
Table 14: UCM6102 Network Settings->Port Forwarding
WAN Port
Specify the WAN port number. Up to 8 ports can be configured.
LAN IP
Specify the LAN IP address.
LAN Port
Specify the LAN port number.
Protocol Type
Select protocol type "UDP Only", "TCP Only" or "TCP/UDP" for the forwarding in
the selected port. The default setting is "UDP Only".
The following figures demonstrate a port forwarding example to provide phone’s web UI access to public
side.
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
The UCM6102 network mode is set to “Route”

The UCM6102 WAN port is connected to uplink switch, with a public IP address configured, e.g.
1.1.1.1.

The UCM6102 LAN port provides DHCP pool that connects to multiple phone devices in the LAN
network 192.168.2.x. The UCM6102 is used as a router, with gateway address 192.168.2.1

There is a GXP2160 connected under the LAN interface network of the UCM6102. It obtains IP
address 192.168.2.100 from UCM6102 DHCP pool

On the UCM6102 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 UCM6102.
Protocol Type: We select TCP here for web UI access using HTTP.
Figure 26: UCM6102 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 27: GXP2160 Web Access Using UCM6102 Port Forwarding
FIREWALL
The UCM6100 provides users firewall configurations to prevent certain malicious attack to the UCM6100
system. Users could configure to allow, restrict or reject specific traffic through the device for security and
bandwidth purpose. The UCM6100 also provides Fail2ban feature for authentication errors in SIP
REGISTER, INVITE and SUBSCRIBE. To configure firewall settings in UCM6100, 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 UCM6100.
Table 15: UCM6100 Firewall->Static Defense->Current Service
Port
Process
Type
Protocol or Service
7777
Asterisk
tcp/IPv4
SIP
389
Slapd
tcp/IPv4
LDAP
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22
Dropbear
tcp/IPv4
SSH
80
Lighthttpd
tcp/IPv4
HTTP
8089
Lighthttpd
tcp/IPv4
HTTPS
69
Opentftpd
udp/IPv4
TFTP
9090
Asterisk
udp/IPv4
SIP
6060
zero_config
udp/IPv4
UCM6100 zero_config service
5060
Asterisk
udp/IPv4
SIP
4569
Asterisk
udp/IPv4
SIP
5353
zero_config
udp/IPv4
UCM6100 zero_config service
37435
Syslogd
udp/IPv4
Syslog
For typical firewall settings, users could configure the following options on the UCM6100.
Table 16: Typical Firewall Settings
Ping Defense
Enable
SYN-Flood Defense
Enable
Ping-of-Death
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 (UCM6102 only) interface.
Enable to prevent SYN Flood denial-of-service attack to the device. The default
setting is disabled. To enable or disable it, click on the check box for the LAN or
WAN (UCM6102 only) interface.
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
(UCM6102 only) interface.
Under "Custom Firewall Settings", users could create new rules to accept, reject or drop certain traffic
going through the UCM6100. To create new rule, click on "Create New Rule" button and a new window will
pop up for users to specify rule options.
Right next to "Create New Rule" button, there is a checkbox for option "Reject Rules". If it’s checked, all
the rules will be rejected except the firewall rules listed below. In the firewall rules, only when there is a rule
that meets all the following requirements, the option "Reject Rules" will be allowed to check:

Action: "Accept"

Type "In"

Destination port is set to the system login port (e.g., by default 8089)

Protocol is not UDP
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Figure 28: Create New Firewall Rule
Table 17: Firewall Rule Settings
Rule Name
Specify the Firewall rule name to identify the firewall rule.
Select the action for the Firewall to perform.
Action

ACCEPT

REJECT

DROP
Select the traffic type.

Type
IN
If selected, users will need specify the network interface "LAN" or "WAN"
(for UCM6102 only) for the incoming traffic.

OUT
Select the service type.
Service

FTP

SSH

Telnet

TFTP

HTTP

LDAP

Custom
If "Custom" is selected, users will need specify Source (IP and port),
Destination (IP and port) and Protocol (TCP, UDP or Both) for the service.
Please note if the source or the destination field is left blank, it will be used
as "Anywhere".
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Save the change and click on "Apply" button. Then submit the configuration by clicking on "Apply
Changes" on the upper right of the web page. The new rule will be listed at the bottom of the page with
sequence number, rule name, action, protocol, type, source, destination and operation. More operations
below:

Click on
to edit the rule

Click on
to delete the rule
DYNAMIC DEFENSE
Dynamic defense is supported on the UCM6102 only. It can blacklist hosts dynamically when the LAN
mode is set to "Route" under web GUI->Settings->Network Settings->Basic Settings page. If enabled,
the traffic coming into the UCM6102 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 UCM6102 firewall,
which will then be displayed in the web page. Please refer to the following table for dynamic defense
options on the UCM6102.
Table 18: UCM6102 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
Periodical Time
Interval
number of TCP connections from a host exceeds the connection threshold
within this period, this host will be added into Blacklist. The valid value is
between 1 and 59 when dynamic defense is turned on. The default setting is
59.
Blacklist Update
Configure the blacklist update time interval (in seconds). The default setting is
Interval
120.
Connection
Threshold
Configure the connection threshold. Once the number of connections from the
same host reaches the threshold, it will be added into the blacklist. The default
setting is 100.
Configure the dynamic defense whitelist.
Dynamic Defense
Whitelist
For example,
192.168.1.3
192.168.1.4
The following figure shows a configuration example like this:
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
If a host at IP address 192.168.40.7 initiates more than 20 TCP connections to the UCM6102 within 1
minute, it will be added into UCM6102 blacklist.

This host 192.168.40.7 will be blocked by the UCM6102 for 300 seconds.

Since IP address 192.168.40.5 is in whitelist, if the host at IP address 192.168.40.5 initiates more than
20 TCP connections to the UCM6102 within 1 minute, it will not be added into UCM6102 blacklist. It
can still establish TCP connection with the UCM6102.
Figure 29: Configure Dynamic Defense
FAIL2BAN
Fail2Ban feature on the UCM6100 provides intrusion detection and prevention for authentication errors in
SIP REGISTER, INVITE and SUBSCRIBE. Once the entry is detected within "Max Retry Duration", the
UCM6100 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 19: 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 UCM6100.
Banned Duration
Max Retry Duration
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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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Configure the number of authentication failures during "Max Retry Duration"
MaxRetry
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 UCM6100.
Configure the listening port number for the service. Currently only 5060 (for UDP)
Protocol
is supported.
Configure the number of authentication failures during "Max Retry Duration"
before the host is banned. The default setting is 10. Please make sure this option
MaxRetry
is properly configured as it will override the "MaxRetry" value under "Global
Settings".
CHANGE PASSWORD
After login 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->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 30 characters.
4. Click on "Save" and the user will be automatically logged out.
5. Once the web page comes back to the login page again, enter the username "admin" and the new
password to login.
LDAP SERVER
The UCM6100 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 UCM6100 user extensions already.
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
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 UCM6100 LDAP server have the same Base DN "dc=pbx,dc=com".
If users have the Grandstream phone provisioned by the UCM6100, the LDAP directory has been set up
on the phone and can be used right away for users to access all phonebooks.
Additionally, users could manually configure the LDAP client settings to manipulate the built-in LDAP
server on the UCM6100. If the UCM6100 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
UCM6100 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 UCM6100.
Figure 30: LDAP Server Configurations
The UCM6100 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.
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Figure 31: Default LDAP Phonebook DN
Figure 32: Default LDAP Phonebook Attributes
LDAP PHONEBOOK
Users could use the default phonebook, edit the default phonebook, add new phonebook, import
phonebook on the LDAP server as well as export phonebook from the LDAP server. The first phonebook
with default phonebook dn "ou=pbx,dc=pbx,dc=com" displayed on the LDAP server page is for extensions
in this PBX. Users cannot add or delete contacts directly. The contacts information will need to be modified
via Web GUI->PBX->Basic/Call Routes->Extensions first. The default LDAP phonebook will then be
updated automatically.
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Figure 33: LDAP Server: LDAP Phonebook

Add new phonebook
A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under "LDAP
Phonebook" section.
Figure 34: Add LDAP Phonebook
Configure the "Phonebook Prefix" first. The "Phonebook DN" will be automatically filled in. For example, if
configuring
"Phonebook
Prefix"
as
"people",
the
"Phonebook
DN"
will
be
filled
with
"ou=people,dc=pbx,dc=com".
Once added, users can select
select
to edit the phonebook attributes and contact list (see figure below), or
to delete the phonebook.
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Figure 35: 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 36: Import Phonebook
The file to be imported must be a CSV file with UTF-8 encoding. Users can open the CSV file with Notepad
and save it with UTF-8 encoding.
Here is how a sample file looks like. Please note “Account Number” and “Phonebook DN” fields are
required. Users could export a phonebook file from the UCM6100 LDAP phonebook section first and use it
as a sample to start with.
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Figure 37: 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 UCM6100
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 UCM6100 LDAP Phonebook, the contacts in the CSV file will be added into the
existing phonebook. If the phonebook DN doesn’t exist on the UCM6100 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
UCM6100.
Figure 38: LDAP Phonebook After Import
As the default LDAP phonebook with DN “ou=pbx,dc=pbx,dc=com” cannot be edited or deleted in LDAP
phonebook section, users cannot import contacts with Phonebook DN field “pbx” if existed in the CSV file.

Export phonebook to your computer from UCM6100 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 UCM6100 again.
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Figure 39: 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
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 UCM6100
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 30: LDAP Server Configurations.
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Figure 40: GXP2200 LDAP Phonebook Configuration
HTTP SERVER
The UCM6100 embedded web server responds to HTTP/HTTPS GET/POST requests. Embedded HTML
pages allow the users to configure the PBX through a Web browser such as Microsoft IE, Mozilla Firefox
and Google Chrome. By default, the PBX can be accessed via HTTPS using Port 8089 (e.g.,
https://192.168.40.50:8089). Users could also change the access protocol and port as preferred under
Web GUI->Settings->HTTP Server.
Table 20: 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
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HTTPS with Port 8089. The default setting is "Enable".
Select HTTP or HTTPS. The default setting is "HTTPS". This is also the
Protocol Type
protocol used for zero config when the end point device downloads the
config file from the UCM6100.
Port
Specify port number to access the HTTP server. The default port number
is 8089.
Once the change is saved, the web page will be redirected to the login page using the new URL. Enter the
username and password to login again.
EMAIL SETTINGS
The Email application on the UCM6100 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 21: Email Settings
TLS Enable
Enable or disable TLS during transferring/submitting your Email to other
SMTP server. The default setting is "Yes".
Select Email type.

MTA: Mail Transfer Agent. The Email will be sent from the configured
domain. When MTA is selected, there is no need to set up SMTP
Type
server for it or no user login is required. However, the Emails sent
from MTA might be considered as spam by the target SMTP server.

Client: Submit Emails to the SMTP server. A SMTP server is required
and users need login with correct credentials.
Domain
Specify the domain name to be used in the Email when using type "MTA".
Server
Specify the SMTP server when using type "Client".
Username
Password
Display Name
Sender
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Username is required when using type "Client". Normally it's the Email
address.
Password to login for the above Username (Email address) is required
when using type "Client".
Specify the display name in the FROM header in the Email.
Specify the sender's Email address.
For example, pbx@example.mycompany.com.
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The following figure shows a sample Email settings on the UCM6100, assuming the Email is using
smtp.gmail.com as the SMTP server.
Figure 41: UCM6100 Email Settings
Once the configuration is finished, click on "Test". In the prompt, fill in a valid Email address to send a test
Email to verify the Email settings on the UCM6100.
TIME SETTINGS
AUTO TIME UPDATING
The current system time on the UCM6100 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 UCM6100 to update time automatically, go to Web GUI->Settings->Time
Settings->Time Auto 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
UCM6100 for the first time to avoid service interrupt after installation and deployment in production.
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Table 22: Time Auto Updating
Specify the URL or IP address of the NTP server for the UCM6100 to
Remote NTP Server
synchronize
the
date
and
time.
The
default
NTP
server
is
ntp.ipvideotalk.com.
If set to "Yes", the UCM6100 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 UCM6100 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 UCM6100 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 UCM6100, go to Web GUI->Settings->Time Settings->Set Time
Manually. The format is YYYY-MM-DD HH:MI:SS.
Figure 42: 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 UCM6100 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 UCM6100, 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 43: Create New Office Time
Table 23: 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 44: 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 UCM6100, 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.
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Figure 45: Create New Holiday
Table 24: Create New Holiday
Name
Specify the holiday name to identify this holiday.
Holiday Memo
Create a note for the holiday.
Month
Select the month for the holiday.
Day
Select the day for the holiday.
Show Advanced Options
Week
Check this option to show advanced options. If selected, please specify
the days as holiday in one week below.
Select the days as holiday in one week.
Enter holiday "Name" and "Holiday Memo" for the new holiday. Then select "Month" and "Day". The
system administrator can also define days in one week as advanced options. Once done, click on "Save"
and then "Apply Change" for the holiday to take effect. The holiday will be listed in the web page as the
figure shows below.
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Figure 46: 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/products/ucm_series/ucm61xx/documents/how_to_use_office_time_and_hol
iday_ucm6100.pdf
NTP SERVER
The UCM6100 can be used as a NTP server for the NTP clients to synchronize their time with. To
configure the UCM6100 as the NTP server, set "Enable NTP server" to "Yes" under web
GUI->Settings->NTP Server. On the client side, point the NTP server address to the UCM6100 IP
address or host name to use the UCM6100 as the NTP server.
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RECORDINGS STORAGE
The UCM6100 supports call recordings automatically or manually and the recording files can be saved in
external storage plugged in the UCM6100 or on the UCM6100 locally. To manage the recording storage,
users can go to UCM6100 web GUI->Settings->Recordings Storage page and select whether to store
the recording files in USB Disk, SD card or locally on the UCM6100.
Figure 47: 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 UCM6100. 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 UCM6100 internal storage.
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
If “USB Disk” or “SD Card” is selected, the recordings will be stored in the corresponding plugged in
external storage device. Please note the options “USB Disk” and “SD Card” will be displayed only if
they are plugged into the UCM6100.
Once “USB Disk” or “SD Card” is selected, click on “OK”. The user will be prompted to confirm to copy
the local files to the external storage device.
Figure 48: 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 49: Recording Storage Category
On the UCM6100, 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.
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-
Queue: Copy the call queue recording files to the external device.
-
All: Copy all recording files to the external device.
LOGIN TIMEOUT SETTINGS
After the user logs in the UCM6100 web UI, the user will be automatically logged out after certain timeout.
This timeout value can be specified under UCM100 web GUI->Settings->Login Timeout Settings page.
The “User Login Timeout” value is in minute and the default setting is 10 minutes. If the user doesn’t make
any operation on web UI within the timeout, the user will be logged out automatically. After that, the web UI
will be redirected to the login page and the user will need to enter username and password to log in.
If set to 0, there is no timeout for the web UI login session and the user will not be automatically logged out.
Figure 50: Login Timeout Settings
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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 UCM6100 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 UCM6100. The settings for this feature can be accessed via Web
UI->PBX->Zero Config.
CONFIGURATION ARCHITECTURE FOR END POINT DEVICE
Started from firmware version 1.0.7.10, the end point device configuration in zero config is divided into the
following three layers with priority from the lowest to the highest:

Global
This is the lowest layer. Users can configure the most basic options that could apply to all Grandstream
SIP devices during provisioning via Zero config.

Model
In this layer, users can define model-specific options for the configuration template.

Device
This is the highest layer. Users can configure device-specific options for the configuration for individual
device here.
Each layer also has its own structure in different levels. Please see figure below. The details for each layer
are explained in sections [GLOBAL CONFIGURATION], [MODEL CONFIGURATION] and [DEVICE
CONFIGURATION].
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Figure 51: Zero Config Configuration Architecture for End Point Device
The configuration options in model layer and device layer have all the option in global layers already, i.e.,
the options in global layer is a subset of the options in model layer and device layer. If an option is set in all
three layers with different values, the highest layer value will override the value in lower layer. For
example, if the user selects English for Language setting in Global Policy and Spanish for Language
setting in Default Model Template, the language setting on the device to be provisioned will use Spanish as
model layer has higher priority than global layer. To sum up, configurations in higher layer will always
override the configurations for the same options/fields in the lower layer when presented at the
same time.
After understanding the zero config configuration architecture, users could configure the available options
for end point devices to be provisioned by the UCM6100 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 UCM6100 for auto provisioning. Three methods of
auto provisioning are used.
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Figure 52: UCM6100 Zero Config

SIP SUBSCRIBE
When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The
UCM6100 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 UCM6100 and take the
new configuration.

DHCP OPTION 66
This method should be used on the UCM6102 because only the UCM6102 has WAN and LAN port with
LAN port supporting the router function. When the phone restarts (by default DHCP Option 66 is turned
on), it will send out a DHCP DISCOVER request. The UCM6102 receives it and returns DHCP OFFER
with the config server path URL in Option 66, for example, https://192.168.2.1:8089/zccgi/. The phone
will then use the path to download the config file generated in the UCM6100.

mDNS
When the phone boots up, it sends out mDNS query to get the TFTP server address. The UCM6100
will respond with its own address. The phone will then send TFTP request to download the XML config
file from the UCM6100.
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 53: Auto Provision Settings
Table 25: 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
UCM6100 will not send the SIP device the URL to download the config file
and therefore the SIP device will not be automatically provisioned by the
Enable Automatic
Configuration Assignment
UCM6100.
Note:
When disabled, SIP devices can still be provisioned by manually sending
NOTIFY from the UCM6100 which will include the XML config file URL for
the SIP device to download.
If enabled, when the device is discovered, the PBX will automatically
Automatically Assign Extension
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 range to be assigned if "Automatically Assign Extension" is
Zero Config Extension
enabled. The default range is 5000-6299. Zero Config Extension
Segment
Segment
range
can
be
defined
in
web
UI->PBX->Internal
Options->General page->Extension Preference section: "Auto Provision
Extensions".
Enable Pick Extension
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If enabled, the extension list will be sent out to the device after receiving
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the device's request. This feature is for the GXP series phones that
support selecting extension to be provisioned via phone's LCD. The
default setting is disabled.
Click on the link "Pick Extension Segment" to specify the extension list to
be sent to the device. The default range is 4000 to 4999. Pick Extension
Pick Extension Segment
Segment
range
Options->General
can
be
defined
page->Extension
in
web
Preference
UI->PBX->Internal
section:
"Pick
Extensions".
Pick Extension Period (hour):
Specify the number of minutes to allow the phones being provisioned to
pick extensions.
Please make sure an extension is manually assigned to the phone or "Automatically Assign Extension" is
enabled during provisioning. After the configuration on the UCM6100 web GUI, click on "Save" and "Apply
Changes". Once the phone boots up and picks up the config file from the UCM6100, 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", 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 UCM6100. If users need scan the entire
network segment, enter 255 (for example, 192.168.40.255) instead of a specific IP address. Then click on
"Save" to start discovering the devices within the same network. To successfully discover the devices,
"Zero Config" needs to be enabled on the UCM6100 web GUI->PBX->Zero Config->Auto Provisioning
Settings.
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Figure 54: Auto Discover
The following figure shows a list of discovered phones. The MAC address, IP Address, Extension (if
assigned), Version, Vendor, Model, Connection Status, Create Config, Options (Edit/Delete/Update) are
displayed in the list.
Figure 55: 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 UCM6100 no matter what the Grandstream device model it is. It is divided into two levels:

Web UI->PBX->Zero Config->Global Policy

Web UI->PBX->Zero Config->Global Templates.

Global Templates configuration has higher priority to Global Policy configuration.
Global Policy can be accessed in web GUI->PBX->Zero Config->Global Policy page. On the top of the
configuration table, users can select category in the "Options" dropdown list to quickly navigate to the
category. The categories are:

Localization: configure display language, data and time.

Phone Settings: configure dial plan, call features, NAT, call progress tones and etc.
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
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 56: 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 26: Global Policy Parameters: Localization
Language settings
Language
Select the LCD display language on the SIP end device.
Date and Time
Date Format
Time Format
NTP Server
Time Zone
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Configure the date display format on the SIP end device’s LCD.
Configure the time display in 12-hour or 24-hour format on the SIP end
device’s 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.
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Table 27: Global Policy Parameters: Phone Settings
Default Call Settings
Dial Plan
Configure the default dial plan rule. For syntax and examples, please
refer to user manual of the SIP devices to be provisioned for more details.
When enabled, “Do Not Disturb”, “Call Forward” and other call features
Enable Call Features
can be used via the local feature code on the phone. Otherwise, the ITSP
feature code will be used.
Use # as Dial Key
If set to “Yes”, pressing the number key “#” will immediately dial out the
input digits.
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.
General Settings
Configure call progress tones including ring tone, dial tone, second dial
tone, message waiting tone, ring back tone, call waiting tone, busy tone
and reorder tone using the following syntax:
f1=val, f2=val[, c=on1/ off1[- on2/ off2[- on3/ off3]]];
Call Progress Tones

Frequencies are in Hz and cadence on and off are in 10ms).

“on” is the period (in ms) of ringing while “off” is the period of silence.
Up to three cadences are supported.

Please refer to user manual of the SIP devices to be provisioned for
more details
Select “Default Mode” or “Toggle Headset/Speaker” for the Headset key.
HEADSET Key Mode
Please refer to user manual of the SIP devices to be provisioned for more
details.
Table 28: Global Policy Parameters: Contact List
LDAP Phonebook
Select "Manual" or "PBX" as the LDAP configuration source.

Source
If "Manual" is selected, the LDAP configuration below will be applied
to the SIP end device.

If "PBX" is selected, the LDAP configuration built-in from UCM6100
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.
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This is the location in the directory where the search is requested to
Base DN
User Name
Password
Number Filter
Name Filter
Version
begin. Example:

dc=grandstream, dc=com

ou=Boston, dc=grandstream, dc=com
Configure the bind "Username" for querying LDAP servers. The field can
be left blank if the LDAP server allows anonymous binds.
Configure the bind "Password" for querying LDAP servers. The field can
be left blank if the LDAP server allows anonymous binds.
Configure the filter used for number lookups. Please refer to user manual
for more details.
Configure the filter used for name lookups. Please refer to user manual
for more details.
Select the protocol version for the phone to send the bind requests. The
default value is 3.
Specify the "name" attributes of each record which are returned in the
Name Attribute
LDAP search result. Example:

gn

cn sn description
Specify the "number" attributes of each record which are returned in the
Number Attribute
LDAP search result. Example:

telephoneNumber

telephoneNumber Mobile
Configure the entry information to be shown on phone's LCD. Up to 3
Display Name
fields 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.
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
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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.
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XML Phonebook
Select the source of the phonebook XML server.

Disable
Disable phonebook XML downloading.

Manual
Once selected, users need specify downloading protocol HTTP,
HTTPS or TFTP and the server path to download the phonebook
Phonebook XML Server
XML file. The server path could be IP address or URL, with up to 256
characters.

Local UCM Server
Once selected, click on the Server Path field to upload the phonebook
XML file. Please note: after uploading the phonebook XML file to the
server, the original file name will be used as the directory name and
the file will be renamed as phonebook.xml under that directory.
Phonebook Download Interval
Configure the phonebook download interval (in Minute). If set to 0,
automatic download will be disabled. Valid range is 5 to 720.
Remove manually-edited
If set to "Yes", when XML phonebook is downloaded, the entries added
entries on download
manually will be automatically removed.
Table 29: 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
UCM6100.
Select a source to get the firmware file:

URL
If select to use URL to upgrade, complete the configuration for the
following four parameters: “Upgrade Via”, “Server Path”, “File Prefix”
Firmware Source
and “File Postfix”.

Local UCM Server
Firmware can be uploaded to the UCM6100 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
UCM6100 and the firmware file must be put under a folder named
“ZC_firmware” in the USB storage root directory.
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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 endpoint device will automatically upgrade if a new
firmware is detected. Users can select automatic upgrading by day, by
week or by minute.

By week
Once selected, specify the day of the week to check HTTP/TFTP
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.
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
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Configure the URL/IP address for the syslog server.
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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 30: 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)
IP Address
used by the client and server to exchange vendor class ID
information.

PPPoE
Once selected, users need specify the Account ID, Password and
Service Name for PPPoE.
Advanced Setting
Layer 3 QoS
Define the Layer 3 QoS parameter. This value is used for IP Precedence,
Diff-Serv or MPLS. Valid range is 0-63.
Layer 2 QoS Tag
Assign the VLAN Tag of the Layer 2 QoS packets. Valid range is 0 -4095.
Layer 2 QoS Priority Value
Assign the priority value of the Layer 2 QoS packets. Valid range is 0-7.
STUN Server
Configure the IP address or Domain name of the STUN server. Only
non-symmetric NAT routers work with STUN.
Specify how often the phone will send a blank UDP packet to the SIP
Keep Alive Interval
server in order to keep the "ping hole" on the NAT router to open. Valid
range is 10-160.
Table 31: Global Policy Parameters: Customization
Wallpaper
Check this option if the SIP end device shall use 1024 x 600 resolution for
the LCD screen wallpaper.

Screen Resolution 1024 x 600
Source
Configure the location where wallpapers are stored.

File
If "URL" is selected as source, specify the URL of the wallpaper file. If
"Local UCM Server" is selected as source, click to upload wallpaper
file to the UCM6100.
Screen Resolution 800 x 400
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Check this option if the SIP end device shall use 800 x 400 resolution for
the LCD screen wallpaper.
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
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 UCM6100.
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 UCM6100.
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 UCM6100.
GLOBAL TEMPLATES
Global Templates can be accessed in web GUI->PBX->Zero Config->Global Templates. Users can
create multiple global templates with different sets of configurations and save the templates. Later on,
when the user configures the device in Edit Device dialog->Advanced Settings, the user can select to use
one of the global template for the device. Please refer to section [MANAGE DEVICES] for more details on
using the global templates.
When creating global template, users can select the categories and the parameters under each categories
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.
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Click on "Create New Template" to add a global template. Users will see the following configurations.
Table 32: Create New Template
Template Name
Create a name to identify this global template.
Description
Provide a description for the global template. This is optional.
Active
Check this option to enable the global template.

Click on
to edit the global template.
The window for editing global template is shown in the following figure. In the “Options” field, after
entering the option name key word, the options containing the key word will be listed. Users could then
select the options to be modified and click on “Add Option” to add it into the global template.
Figure 57: Edit Global Template
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The added options will show in the list. Users can then enter or select value for each option to be used
in the global template. On the left side of each added option, users can click on
option from the template. On the right side of each option, users can click on
value to the default value.
to remove this
to reset the option
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.
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 UCM6100.
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.
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Table 33: Create New Model Template
Select a model to apply this template. The supported Grandstream
Model
models are listed in the dropdown list for selection.
Template Name
Create a name for the model template.
Description
Enter a description for the model template. This is optional.
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.
Active

Check this option to enable the model template.
Click on
to edit the model template.
The editing window for model template is shown in the following figure. In the “Options” field, enter the
option name key word, the option that contains the key word will be listed. User could then select the
option and click on “Add Option” to add it into the model template.
Once added, the option will be shown in the list below. On the left side of each option, users can click
on
to remove this option from the model template. On the right side of each option, users can click
on
to reset the option to the default value.
User could also click on “Add New Field” to add a P value number and the value to the configuration.
The following figure shows setting P value “P1362” to “en”, which means the display language on the
LCD is set to English. For P value information of different models, please refer to configuration
template here http://www.grandstream.com/support/tools.
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Figure 58: Edit Model Template

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
UCM6100 zero config feature supports provisioning all models of Grandstream SIP end devices.
Templates for most of the Grandstream models are built in with the UCM6100 already. Templates for
GSWave and Surveillance products requires user to download on UCM6100 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 UCM6100.
Figure 59: Template Management
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 UCM6100. 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.
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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 60: Create New Device
MANAGE DEVICES
The device manually created or discovered from Auto Discover will be listed in the web UI->PBX->Zero
Config->Zero Config page. Users can see the devices with their MAC address, IP address, vendor,
model and etc.
Figure 61: Manage Devices

Click on
to access the web UI of the phone.
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
Click on
to edit the device configuration.
A new dialog will be displayed for the users to configure “Basic” settings and “Advanced” settings.
“Basic” settings have the same configurations as displayed when manually creating a new device, i.e.,
account, line key and MPK settings; “Advanced” settings allow users to configure more details in a
five-level structure.
Figure 62: Edit Device
A preview of the “Advanced” settings is shown in the above figure. There are five levels configurations
as described in (1) (2) (3) (4) (5) below, with priority from the lowest to the highest. The configurations
in all levels will take effect for the device. If there are same options existing in different level
configurations with different value configured, the higher level configuration will override the lower level
configuration.
(1) Global Policy
This is the lowest level configuration. The global policy configured in web UI->PBX->Zero
Config->Global Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page
PBX->Zero Config->Global Policy.
(2) Global Templates
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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 33: Create New Model Template].
(4) Model Templates
Select a model template to be used for the device and click on
to add. Multiple global
templates can be selected and users can arrange the priority by adjusting orders via
and
.
All the selected model templates will take effect. If the same option exists on multiple selected
model templates, the value in the template with higher priority will override the one in the template
with lower priority. Click on
to remove the model template from the selected list.
(5) Customize Device Settings
This is the highest level configuration for the device. Click on “Modify Customize Device Settings”
and following dialog will show.
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Figure 63: Edit Customize Device Settings
Scroll down in the dialog to view and edit the device-specific options. If the users would like to
add more options which are not in the pre-defined list, click on “Add New Field” to add a P value
number and the value to the configuration. The following figure shows setting P value “P1362” to
“en”, which means the display language on the LCD is set to English. The warning information on
right tells that the option matching the P value number exists and clicking on it will lead to the
matching option. For P value information of different models, please refer to configuration
template here http://www.grandstream.com/support/tools.
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Figure 64: 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 65: 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 66: Modify Selected Devices - Different Models
Note:
Performing batch operation will override all the existing device configuration on the page.
After the above configurations, save the changes and go back to web UI->PBX->Zero Config->Zero
Config page. Users could then click on
to send NOTIFY to the SIP end point device and trigger the
provisioning process. The device will start downloading the generated configuration file from the URL
contained in the NOTIFY message.
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Figure 67: 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 68: 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 69: 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 70: Zero Config Sample - Device Preview 2
4 “Model Template”. The preview of the device settings is
Select “Spanish Support Template” in ○
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 71: 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 UCM6100.
Now all the 9 phones in the network will be provisioned with an unique extension registered on the
UCM6100. 3 of the phones will be provisioned to display Spanish on LCD and the other 5 will be
provisioned to display English on LCD. The GXV3275 used by the supervisor will be provisioned to use the
default language on LCD display since it’s not specified in the global policy.
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EXTENSIONS
CREATE NEW USER
CREATE NEW SIP EXTENSION
To manually create new SIP user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on
"Create New User"->"Create New SIP Extension" and a new dialog window will show for users to fill in the
extension information.
Figure 72: 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.
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The configuration parameters are as follows.
Table 34: 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 UCM6100.
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
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Configure the first name of the user. The first name can contain
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characters, letters, digits and _.
Last Name
Email Address
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.
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
Language
shows all the current available voice prompt languages on the UCM6100.
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 35: SIP Extension Configuration Parameters: Media
SIP Settings
Use NAT when the UCM6100 is on a public IP communicating with
NAT
devices hidden behind NAT (e.g., broadband router). If there is one-way
audio issue, usually it's related to NAT configuration or Firewall's support
of SIP and RTP ports. The default setting is enabled.
By default, the UCM6100 will route the media steams from SIP endpoints
through itself. If enabled, the PBX will attempt to negotiate with the
Can Reinvite
endpoints to route the media stream directly. It is not always possible for
the UCM6100 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.

Port: Allow peers matching by IP address without matching port
number.

Very: Allow peers matching by IP address without matching port
number. Also, authentication of incoming INVITE messages is not
Insecure
required.

No: Normal IP-based peers matching and authentication of incoming
INVITE.
The default setting is "Port".
Enable Keep-alive
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If enabled, empty SDP packet will be sent to the SIP server periodically to
keep the NAT port open. The default setting is "Yes".
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Keep-alive Frequency
Configure the Keep-alive interval (in seconds) to check if the host is up.
The default setting is 60 seconds.
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
TEL URI
be attached to the Request-Line and TO header in the SIP request to
indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of
"SIP:" in the SIP request. The default setting is disabled.
SRTP
Enable SRTP for the call. The default setting is disabled.
Select Fax mode. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
Fax Mode
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web 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:
Codec Preference
PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, ILBC,
ADPCM, H.264, H.263 and H.263p.
Table 36: SIP Extension Configuration Parameters - Features
Call Transfer
Configure the Call Forward Unconditional target number. If not
Call Forward Unconditional
configured, the Call Forward Unconditional feature is deactivated. The
default setting is deactivated.
Select time condition for Call Forward Unconditional. CFU takes effect
CFU Time Condition
only during the selected time condition. The available time condition are
“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
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Note:

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
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
condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday” and “Specific”.
Note:

CFN Time Condition
“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page 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 condition
are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:

CFB Time Condition
“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
CC Settings
If enabled, UCM6100 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.
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Two modes for Call Completion are supported:
CC Mode

Normal: This extension is used as ordinary extension.

For Trunk: This extension is registered from a PBX.
The default setting is “Normal”.
Configure the maximum number of CCSS agents which may be allocated
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.
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
UCM6100, which can be configured in the global ring timeout setting
Ring Timeout
under web GUI->Internal Options->IVR Prompt: General Preference. The
valid range is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
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.
Music On Hold
Skip Trunk Auth
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Select which Music On Hold class to suggest to extension when putting it
on hold.
If enabled, users will not need enter the "PIN Set" required by the
outbound rule to make outbound calls. The default setting is "No".
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Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Table 37: SIP Extension Configuration Parameters - Specific Time
Specific Time
Time Condition
Click to add Time Condition to configure specific time for this extension.
CREATE NEW IAX EXTENSION
To manually create new IAX user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on
"Create New User"->"Create New IAX Extension" and a new dialog window will show for users to fill in the
extension information. The configuration parameters are as follows.
Table 38: 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.
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
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When user dials voicemail code, the password verification IVR is skipped.
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
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If selected, this extension will be disabled on the UCM6100.
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.
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
Language
shows all the current available voice prompt languages on the UCM6100.
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 39: 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".
SRTP
Enable SRTP for the call. The default setting is disabled.
Select Fax Mode. The default setting is “None”.
Fax Mode

None: Disable Fax. This is the default setting.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Strategy
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This option controls how the extension can be used on devices within
different types of network.
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
Allow All
Device in any network can register this extension.

Local Subnet Only
Only the user in specific subnet can register this extension. Up to
three subnet addresses can be specified.

A Specific IP Address
Only the device on the specific IP address can register this extension.
The default setting is "Allow All".
Select audio and video codec for the extension. The available codecs are:
Codec Preference
PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729, G.723, ILBC,
ADPCM, H.264, H.263 and H.263p.
Table 40: IAX Extension Configuration Parameters - Features
Call Transfer
Configure the Call Forward Unconditional target number. If not
Call Forward Unconditional
configured, the Call Forward Unconditional feature is deactivated. The
default setting is deactivated.
Select time condition for Call Forward Unconditional. CFU takes effect
only during the selected time condition. The available time condition 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
condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday” and “Specific”.
CFN Time Condition
Note:

“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Firmware Version 1.0.8.12
Specific time can be configured on the bottom of the extension
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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 condition
are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:

CFB Time Condition
“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page 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
forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6100, which can be configured in the global ring timeout setting
Ring Timeout
under web GUI->Internal Options->IVR Prompt: General Preference. The
valid range is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
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.
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Select which Music On Hold class to suggest to extension when putting it
Music On Hold
on hold.
If enabled, users will not need enter the "PIN Set" required by the
Skip Trunk Auth
outbound rule to make outbound calls. The default setting is "No".
Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Table 41: 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
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.
Table 42: 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.
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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 UCM6100.
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.
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
Language
shows all the current available voice prompt languages on the UCM6100.
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 43: 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.
MAX RX Flash
Firmware Version 1.0.8.12
Configure the maximum period of time (in milliseconds) that the
hook-flash must remain unpressed for the PBX to consider the event as a
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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
Enable Polarity Reversal
polarity reversal is used for signaling the disconnection of a phone line
and the call will be considered as hangup on a polarity reversal. The
default setting is "Yes".
Specify "ON", "OFF" or a value (the power of 2) from 32 to 1024 as the
number of taps of cancellation.
Echo Cancellation
Note:
When configuring the number of taps, the number 256 is not translated
into 256ms of echo cancellation. Instead, 256 taps means 256/8 = 32 ms.
The default setting is "ON", which is 128 taps.
3-Way Calling
Send CallerID After
Configure to enable/disable 3-way calling feature on the user. The default
setting is enabled.
Configure the number of rings before sending CID. Default setting is 1.
For FXS extension, there are three options available in Fax Mode. The
default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
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.

Fax Gateway: If selected, the UCM6100 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 44: FXS Extension Configuration Parameters - Features
Call Transfer
Configure the Call Forward Unconditional target number. If not
Call Forward Unconditional
configured, the Call Forward Unconditional feature is deactivated. The
default setting is deactivated.
Select time condition for Call Forward Unconditional. CFU takes effect
only during the selected time condition. The available time condition are
CFU Time Condition
“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:

Firmware Version 1.0.8.12
“Specific” has higher priority to “Office Times” if there is a conflict in
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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
condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of
Holiday”, “Out of Office Time or Holiday” and “Specific”.
Note:

CFN Time Condition
“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page 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 condition
are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of
Office Time or Holiday” and “Specific”.
Note:

CFB Time Condition
“Specific” has higher priority to “Office Times” if there is a conflict in
terms of time period.

Specific time can be configured on the bottom of the extension
configuration dialog. Scroll down the add Time Condition for specific
time.

Office Time and Holiday could be configured on page Settings->Time
Settings->Office Time/Holiday page.
CC Settings
If enabled, UCM6100 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
Ring Simultaneously
Firmware Version 1.0.8.12
Enable this option to have an external number ring simultaneously along
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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.
Set the external number to be rang simultaneously. ‘-’ is the connection
External Number
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
UCM6100, which can be configured in the global ring timeout setting
Ring Timeout
under web GUI->Internal Options->IVR Prompt: General Preference. The
valid range is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
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.
Select which Music On Hold class to suggest to extension when putting it
Music On Hold
on hold.
If enabled, users will not need enter the "PIN Set" required by the
Skip Trunk Auth
outbound rule to make outbound calls. The default setting is "No".
Dial Trunk Password
Configure personal password when making outbound calls via trunk.
Table 45: 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
Under Web GUI->PBX->Basic/Call Routes->Extensions, click on "Batch Add Extensions"->"Batch Add
SIP Extensions".
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Table 46: 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
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 from this rule.
Enable Voicemail
Enable Voicemail for the user. The default setting is "Yes".
Configure the SIP/IAX password for the users. Three options are
available to create password for the batch of extensions.

SIP/IAX Password
User Random Password.
A random secure password will be automatically generated. It is
recommended to use this password for security purpose.

Use Extension as Password.

Enter a password to be used on all the extensions in the batch.
Configure Voicemail password (digits only) for the users.

User Random Password.
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
UCM6100, which can be configured in the global ring timeout setting
Ring Timeout
under web GUI->Internal Options->IVR Prompt: General Preference.
The valid range is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
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
Firmware Version 1.0.8.12
When user dials voicemail code, the password verification IVR is skipped.
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Verification
If enabled, this would allow one-button voicemail access. By default this
option is disabled.
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 itself. If enabled, the PBX will attempt to negotiate with the
Can Reinvite
endpoints to route the media stream directly. It is not always possible for
the PBX to negotiate endpoint-to-endpoint media routing. The default
setting is "No".
Select DTMF mode for the user to send DTMF. The default setting is
"RFC2833". If "Info" is selected, SIP INFO message will be used. If
DTMF Mode
"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.

Port: Allow peers matching by IP address without matching port
number.

Very: Allow peers matching by IP address without matching port
number. Also, authentication of incoming INVITE messages is not
Insecure
required.

No: Normal IP-based peers matching and authentication of incoming
INVITE.
The default setting is "Port".
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
TEL URI
be attached to the Request-Line and TO header in the SIP request to
indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of
"SIP:" in the SIP request. The default setting is disabled.
Other Settings
SRTP
Enable SRTP for the call. The default setting is "No".
Select Fax mode for this user. The default setting is “None”.
Fax Mode
Firmware Version 1.0.8.12

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
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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.
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".
Skip Trunk Auth
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.
BATCH ADD IAX EXTENSIONS
Under Web GUI->PBX->Basic/Call Routes->Extensions, click on "Batch Add Extensions"->"Batch Add
IAX Extensions".
Table 47: 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
"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 from this rule.
Enable Voicemail
Firmware Version 1.0.8.12
Enable Voicemail for the user. The default setting is "Yes".
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Configure the SIP/IAX password for the users. Three options are
available to create password for the batch of extensions.

SIP/IAX Password
User Random Password.
A random secure password will be automatically generated. It is
recommended to use this password for security purpose.

Use Extension as Password.

Enter a password to be used on all the extensions in the batch.
Configure Voicemail password (digits only) for the users.

User Random Password.
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
UCM6100, which can be configured in the global ring timeout setting
Ring Timeout
under web GUI->Internal Options->IVR Prompt: General Preference.
The valid range is between 5 seconds and 600 seconds.
Note:
If the end point also has a ring timeout configured, the actual ring timeout
used is the shortest time set by either device.
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
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.
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
Fax Mode
Firmware Version 1.0.8.12
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.
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 enabled, users will not need enter the "PIN Set" required by the
Skip Trunk Auth
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.
EDIT EXTENSION
All the UCM6100 extensions are listed under Web GUI->PBX->Basic/Call Routes->Extensions, with
status, Extension, CallerID Name, Technology (SIP, IAX and FXS), IP and Port. Each extension has a
checkbox for users to "Modify Selected Extensions" or "Delete Selected Extensions". Also, options "Edit"
, "Reboot"

and "Delete"
are available per extension.
Status
Users can see the following icon for each extension to indicate the SIP status.
Green:
Free
Blue:
Ringing
Yellow:
In Use
Grey:
Unavailable (the extension is not registered or disabled on the PBX)
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
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 UCM6100 extension already
registered. To successfully reboot the user, "Zero Config" needs to be enabled on the UCM6100 web
GUI->PBX->Zero Config->Auto Provisioning Settings.

Delete single extension
Click on
to delete the extension. Or select the checkbox of the extension and then click on "Delete
Selected Extensions".

Modify selected extensions
Select the checkbox for the extension(s). Then click on "Modify Selected Extensions" to edit the
extensions in a batch.

Delete selected extensions
Select the checkbox for the extension(s). Then click on "Delete Selected Extensions" to delete the
extension(s).
EXPORT EXTENSIONS
The extensions configured on the UCM6100 can be exported to csv format file with selected technology
"SIP", "IAX" or "FXS". Click on "Export Extensions" button and select technology in the prompt below.
Figure 73: Export Extensions
The exported csv file can be serve as a template for users to fill in desired extension information to be
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imported to the UCM6100.
IMPORT EXTENSIONS
The capability to import extensions to the UCM6100 provides users flexibility to batch add extensions with
similar or different configuration quickly into the PBX system.
1. Export extension csv file from the UCM6100 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 74: Import Extensions
4. Select the option in "On Duplicate Extension" to define how the duplicate extension(s) in the imported
csv file should be treated by the PBX.

Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension
information as previously configured without change.

Delete and Recreate: The current extension previously configured will be deleted and the
duplicate extension in the csv file will be loaded to the PBX.

Update Information: The current extension previously configured in the PBX will be kept. However,
if the duplicate extension in the csv file has different configuration for any options, it will override
the configuration for those options in the extension.
5. Click on
to select csv file from local directory in the PC.
6. Click on "Save" to import the csv file.
7. Click on "Apply Changes" to apply the imported file on the UCM6100.
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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 UCM6100
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 75: Email To User - Prompt Information
The user will receive Email including account registration information and LDAP configuration. A QR code
is also generated for Mobile applications to scan it and get automatically provisioned. QR code
provisioning is supported on Grandstream Softphone GS Wave AndroidTM application.
Figure 76: Account Registration Information and QR Code
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Figure 77: LDAP Client Information and QR Code
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TRUNKS
ANALOG TRUNKS
Go to Web GUI->PBX->Basic/Call Routes->Analog Trunks to add and edit analog trunks.

Click on "Create New Analog Trunk" to add a new analog trunk.

Click on
to edit the analog trunk.

Click on
to delete the analog trunk.
ANALOG TRUNK CONFIGURATION
The analog trunk options are listed in the table below.
Table 48: Analog Trunk Configuration Parameters
Select the channel for the analog trunk.
Channels
Trunk Name

UCM6102: 2 channels

UCM6104: 4 channels

UCM6108: 8 channels

UCM6116: 16 channels
Specify a unique label to identify the trunk when listed in outbound rules,
incoming rules and etc.
Enable this option to satisfy two primary use cases, which include
SLA Mode
emulating 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
Barge Allowed
join a call in progress on this trunk. If enabled, the other stations can
press the line button to join the call. The default setting is Yes.
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
Enable Polarity Reversal
Firmware Version 1.0.8.12
If enabled, a polarity reversal will be marked as received when an
outgoing call is answered by the remote party. For some countries, a
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polarity reversal is used for signaling the disconnection of a phone line
and the call will be considered as “hangup” on a polarity reversal. The
default setting is “No”.
When FXO port answers the call, FXS may send a Polarity Reversal. If
Polarity on Answer Delay
this interval is shorter than the value of “Polarity on Answer Delay”, the
Polarity Reversal will be ignored. Otherwise, the FXO will onhook to
disconnect the call. The default setting is 600ms.
Current Disconnect Threshold
(ms)
This is the periodic time (in ms) that the UCM6100 will use to check on a
voltage drop in the line. The default setting is 200. The valid range is 50 to
3000.
Configure the ring timeout (in ms). Trunk (FXO) devices must have a
timeout to determine if there was a hangup before the line is answered.
Ring Timeout
This value can be used to configure how long it takes before the
UCM6100 considers a non-ringing line with hangup 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”.
There are three options available in Fax Mode. The default setting is
“None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
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.

Fax Gateway: If selected, the UCM6100 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.
Caller ID Scheme
FXO Dial Delay(ms)
Select the Caller ID scheme for this trunk. The default setting is
“Bellcore/Telcordia”.
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
Auto Record
setting is disabled. The recording files can be accessed under web
GUI->CDR->Recording Files.
Disable This Trunk
Firmware Version 1.0.8.12
If selected, the trunk will be disabled.
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This is to implement analog trunk outbound line selection strategy. Three
options are available:

Ascend
When the call goes out from this analog trunk, it will always try to use
the first idle FXO port. The port order that the call will use to go out
would be port 1->port 2->port 10->port 16. Every time it will start with
port 1 (if it's idle).

Poll
When the call goes out from this analog trunk, it will use the port that
DAHDI Out Line Selection
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 hangup or for detecting busy
signal. The default setting is "Yes".
If "Busy Detection" is enabled, users can specify the number of busy
tones to be played before hanging up. The default setting is 2. Better
Busy Tone Count
results might be achieved if set to 4, 6 or even 8. Please note that the
higher the number is, the more time is needed to hangup the channel.
However, this might lower the probability to get random hangup.
Congestion Detection
Congestion 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]];
Busy Tone
Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000)
Busy Level Range: (-300, 0)
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Cadence Range: [0, 16383].
Select Tone Country "Custom" to manually configure Busy Tone value.
Default value:
f1=480@-50,f2=620@-50,c=500/500
Syntax:
f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];
Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000)
Congestion Tone
Busy Level Range: (-300, 0)
Cadence Range: [0, 16383].
Select Tone Country "Custom" to manually configure Busy Tone value.
Default value:
f1=480@-50,f2=620@-50,c=250/250
Click on "Detect" to detect the busy tone, Polarity Reversal and Current
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 UCM6100 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
UCM6100 settings.
1. Go to UCM6100 web GUI->PBX->Basic/Call Routes->Analog Trunks page.
2. Click to edit the analog trunk created for the FXO port.
3. In the dialog window to edit the analog trunk, go to "Tone Settings" section and there are two methods
to set the busy tone.

Tone Country. The default setting is "United States of America (USA)".

PSTN Detection.
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Figure 78: UCM6100 FXO Tone Settings
4. Click on "Detect" to start PSTN detection.
Figure 79: UCM6100 PSTN Detection

If there are two FXO ports connected to PSTN lines, use the following settings for auto-detection.
Detect Model: Auto Detect.
Source Channel: The source channel to be detected.
Destination Channel: The channel to help detecting. For example, the second FXO port.
Destination Number: The number to be dialed for detecting. This number must be the actual
PSTN number for the FXO port used as the destination channel.
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Figure 80: UCM6100 PSTN Detection: Auto Detect

If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.
Figure 81: UCM6100 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 UCM6100 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
UCM6100 settings.
Table 49: PSTN Detection for Analog Trunk
Detect Model
Firmware Version 1.0.8.12
Select "Auto Detect" or "Semi-auto Detect" for PSTN detection.
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
Auto Detect
Please make sure two or more channels are connected to the
UCM6100 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
UCM6100 will control the call to be established and hang up between
caller and callee to finish the detection.

Semi-auto Detect
Semi-auto detection requires answering or hanging up the call
manually. Please make sure one channel is connected to the
UCM6100 and in idle status before starting the detection. During the
detection, source channel will be used as caller and send the call to
the configured Destination Number. Users will then need follow the
prompts in web GUI to help finish the detection.
The default setting is "Auto Detect".
Source Channel
Select the channel to be detected.
Destination Channel
Select the channel to help detect when "Auto Detect" is used.
Destination Number
Configure the number to be called to help the detection.
Note:

The PSTN detection process will keep the call up for about 1 minute.

If "Semi-auto Detect' is used, please pick up the call only after informed from the web GUI prompt.

Once the detection is successful, the detected parameters "Busy Tone", "Polarity Reversal" and
"Current Disconnect by PSTN" will be filled into the corresponding fields in the analog trunk
configuration.
VOIP TRUNKS
VoIP trunks can be configured in UCM6100 under Web GUI->PBX->Basic/Call Routes->VoIP Trunks.
Once created, the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and
Options to edit/detect the trunk.

Click on "Create New SIP Trunk" or "Create New IAX Trunk" to add a new VoIP trunk.

Click on
to configure detailed parameters for the VoIP trunk.
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
Click on
to configure Direct Outward Dialing (DOD) for the SIP Trunk.

Click on
to start LDAP Sync.

Click on
to delete the VoIP trunk.
The VoIP trunk options are listed in the table below.
Table 50: Create New SIP Trunk
Select the VoIP trunk type.
Type
Provider Name
Host Name
Keep Trunk CID
Disable This Trunk

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.
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.
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.
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.
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Table 51: SIP Register Trunk Configuration Parameters
Basic Settings
Provider Name
Host Name
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules 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 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.
Disable This Trunk
If selected, the trunk will be disabled.
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
Auto Record
Firmware Version 1.0.8.12
Select whether the trunk needs to register on the external server or not
when "Register SIP Trunk" type is selected. The default setting is No.
Enter the username to register to the trunk from the provider when
"Register SIP Trunk" type is selected.
Enter the password to register to the trunk from the provider when
"Register SIP Trunk" is selected.
Enter the Authentication ID for "Register SIP Trunk" type.
Enable automatic recording for the calls using this trunk (for SIP trunk
only). The default setting is disabled. The recording files can be accessed
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under web GUI->CDR->Recording Files.
Advanced Settings
Select audio and video codec for the VoIP trunk. The available codecs
Codec Preference
are: 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.UCM6100.provider.com" is the From Domain in From
Header: sip:1234567@trunk.UCM6100.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.UCM6100.provider.com.
Send PPI Header
Outbound Proxy Support
Outbound Proxy
DID Mode
If enabled, the INVITE message sent to the trunk will contain PPI
(P-Preferred-Identity) header. The default setting is “No”.
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.

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 UCM6100 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.
Qualify Frequency
Firmware Version 1.0.8.12
When "Enable Qualify" option is set to "Yes", configure the interval (in
seconds) of the SIP OPTIONS message sent to the device to check if the
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device is still online. The default setting is 60 seconds.
Maximum Number of Call Lines
The maximum number of concurrent calls using the trunk. The default
settings 0, which means no limite.
Select Fax mode. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
Fax Mode
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web 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.
Table 52: 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 Trunk CID
Disable This Trunk
TEL URI
Firmware Version 1.0.8.12
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.
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
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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.
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
Codec Preference
are: 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 UCM6100 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 Lines
The maximum number of concurrent calls using the trunk. The default
settings 0, which means no limite.
Select Fax mode. The default setting is “None”.

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
Fax Mode
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
SRTP
Enable SRTP for the VoIP trunk. The default setting is "No".
If enabled, the local UCM6100 will automatically provide and update the
local LDAP contacts to the remote UCM6100 SIP peer trunk. In order to
Sync LDAP Enable
ensure successful synchronization, the remote UCM6100 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
UCM6100 peers o ensure successful synchronization.
Sync LDAP Port
Configure the TCP port used LDAP sync feature between two peer
UCM6100.
Specify an outbound rule for LDAP sync feature. The UCM6100 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 UCM6100 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.
Table 53: Create New IAX Trunk
Select the VoIP trunk type.
Type
Provider Name
Firmware Version 1.0.8.12

Peer IAX Trunk

Register IAX Trunk
Configure a unique label to identify this trunk when listed in outbound
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rules, inbound rules and etc.
Host Name
Keep Trunk CID
Username
Password
Disable This Trunk
Configure the IP address or URL for the VoIP provider’s server of the
trunk.
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
Enter the username to register to the trunk from the provider when
"Register IAX Trunk" type is selected.
Enter the password to register to the trunk from the provider when
"Register IAX Trunk" type is selected.
If selected, the trunk will be disabled.
Table 54: 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
Codec Preference
Firmware Version 1.0.8.12
Select audio and video codec for the VoIP trunk. The available codecs
are: PCMU, PCMA, GSM, AAL2-G.726-32, G.726, G.722, G.729, G.723,
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ILBC, ADPCM, H.264, H.263, H.263p.
Enable Qualify
If enabled, the UCM6100 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 Lines
The maximum number of concurrent calls using the trunk. The default
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
Fax Mode
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Table 55: 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.

Firmware Version 1.0.8.12
If the above two are missing, the "Global Outbound CID" defined in
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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
Codec Preference
are: 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 UCM6100 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 Lines
The maximum number of concurrent calls using the trunk. The default
settings 0, which means no limited.
Select Fax mode. The default setting is “None”.
Fax Mode

None: Disable Fax.

Fax Detect: Fax signal from the user/trunk during the call can be
detected and the received Fax will be sent to the Email address
configured for this extension. If no Email address can be found for the
user, the Fax will be sent to the default Email address configured in
Fax setting page under web UI->PBX->Internal Options->Fax/T.38.
Direct Outward Dialing (DOD)
The UCM6100 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.
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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 UCM6100:
1. To setup DOD go to UCM6100 web GUI->PBX->Basic/Call Routes->VoIP Trunks page.
2. Click
to access the DOD options for the selected SIP Trunk.
3. Click "Create a new DOD" to begin your DOD setup
4. For "DOD Number" enter one of the numbers (DIDs) from your SIP trunk provider. In the example
above Company ABC received 4 DIDs from their provider. ABC will enter in the number for the CEO's
direct line.
5. Select an extension from the "Available Extensions" list. Users have the option of selecting more than
one extension. In this case, Company ABC would select the CEO's extension. After making the
selection, click on the
button to move the extension(s) to the "Selected Extensions" list.
Figure 82: 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.
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Figure 83: Edit DOD
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SLA STATION
Starting from 1.0.7.10, the UCM6100 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 UCM6100 is similar to BLF but SLA is used to monitor
external line i.e., analog trunk on the UCM6100. 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
UCM6100.
CREATE/EDIT SLA STATION
SLA Station can be configured on web GUI->PBX->Basic/Call Routes->SLA Station.
Figure 84: 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 56: SLA Station Configuration Parameters
Station Name
Configure a name to identify the SLA Station.
Station
Specify a SIP extension as a station that will be using SLA.
Available SLA Trunks
Existing Analog Trunks with SLA Mode enabled will be listed here.
Select a trunk for this SLA from the Available SLA Trunks list. Click on
Selected SLA Trunks
to arrange the order. If there are multiple trunks selected, when
there are calls on those trunks at the same time, pressing the LINE key on
the phone will pick up the call on the first trunk here.
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SLA Station Options
Configure the time (in seconds) to ring the station before the call is
Ring Timeout
considered unanswered. No timeout is set by default. If set to 0, there will
be no timeout.
Configure the time (in seconds) for delay before ringing the station when
Ring Delay
a call first coming in on the shared line. No delay is set by default. If set to
0, there will be no delay.
This option defines the competence of the hold action for one particular
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 UCM6100, go to web UI->Basic/Call Routes->Analog Trunks page. Create analog trunk or
edit the existing analog trunk. Make sure “SLA Mode” is enabled for the analog trunk. Once enabled,
this analog trunk will be only available for the SLA stations created under web UI->Basic/Call
Routes->SLA Station page.
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Figure 85: Enable SLA Mode for Analog Trunk
Click on “Save”. The analog trunk will be listed with trunk mode “SLA”.
Figure 86: Analog Trunk with SLA Mode Enabled
2. On the UCM6100, go to web UI->Basic/Call Routes->SLA Station page, click on “Create New SLA
Station”. Please refer to section [CREATE/EDIT SLA STATION] for the configuration parameters. Users
can create one or more SLA stations to monitor the analog trunk. The following figure shows two
stations, 1002 and 1005, are configured to be associated with SLA trunk “fxo1”.
Figure 87: SLA Example - SLA Station
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3. On the SIP phone 1, configure to register UCM6100 extension 1002. Configure the MPK as BLF mode
and the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case.
4. On the SIP phone 2, configure to register UCM6100 extension 1005. Configure the MPK as BLF mode
and value must be set to “extension_trunkname”, which is 1005_fxo1 in this case.
Figure 88: SLA Example - MPK Configuration
Now the SLA station is ready to use. The following functions can be achieved by this configuration.

Making an outbound call from the station/extension, using LINE key
When the extension is in idle state, pressing the line key for this extension on the phone to off hook.
Then dial the station’s extension number, for example, dial 1002 on phone 1 (or dial 1005 on phone 2),
to hear the dial tone. Then the users could dial external number for the outbound call.

Making an outbound call from the station/extension, using BLF key
When the extension is in idle state, pressing the MPK and users could dial external numbers directly.

Answering call using LINE key
When the station is ringing, pressing the LINE key to answer the incoming call.

Barging-in active call using BLF key
When there is an active call between an SLA station and an external number using the SLA trunk,
other SLA stations monitoring the same trunk could join the call by pressing the BLF key if “Barge
Allowed” is enabled for the analog trunk.

Hold/Unhold using BLF key
If the external line is previously put on hold by an SLA station, another station that monitors the same
SLA trunk could unhold the call by pressing the BLF key if “Hold Access” is set to “open” on the analog
trunk and the SLA station.
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CALL ROUTES
OUTBOUND ROUTES
In the UCM6100, 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 UCM6100, the outbound route priority is based on “Best matching pattern”. For example, the
UCM6100 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 57: Outbound Route Configuration Parameters
Calling Rule Name
Configure the name of the calling rule (e.g., local, long_distance, and etc).
Letters, digits, _ and - are allowed.

All patterns are prefixed with the "_".

Special characters:
X: Any Digit from 0-9.
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
Call Duration Limit
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Configure the password for users to use this rule when making outbound
calls.
Enable to configure the maximum duration for the call using this outbound
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route.
Maximum Call Duration
Configure the maximum duration of the call (in seconds). The default
setting is 0, which means no limite.
Configure the warning time for the call using this outbound route. If set to
Warning Time
x seconds, the warning tone will be played to the caller when x seconds
are left to end the call.
Configure the warning repeat interval for the call using this outbound
Warning Repeat Interval
route. If set to x seconds, the warning tone will be played every x seconds
after the first warning.
Select privilege level for the outbound rule.

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.

Disable: The default setting is "Disable". If selected, only the matched
source caller ID will be allowed to use this outbound route.
Please be aware of the potential security risks when using "Internal" level,
which means all users can use this outbound rule to dial out from the
trunk.
When enabled, users could specify extensions allowed to use this
outbound route. "Privilege Level" is automatically disabled if using
"Enable Filter on Source Caller ID".
The following two methods can be used at the same time to define the
extensions as the source caller ID.
1. Select available extensions/extension groups from the left to the right.
This allows users to specify arbitrary single extensions available in
Enable Filter on Source Caller
ID
the PBX.
2. Custom Dynamic Route: define the pattern for the source caller ID.
This allows users to define extension range instead of selecting them
one by one.

All patterns are prefixed with the "_".

Special characters:
X: Any Digit from 0-9.
Z: Any Digit from 1-9.
N: Any Digit from 2-9.
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".": Wildcard. Match one or more characters.
"!": Wildcard. Match zero or more characters immediately.
Example: [12345-9] - Any digit from 1 to 9.
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
Failover Trunk
placed via the regular trunk may have a secondary trunk to go through.
Example:
The user's primary trunk is a VoIP trunk and the user would like to use the
PSTN when the VoIP trunk is not available. The PSTN trunk can be
configured as the failover trunk of the VoIP trunk.
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.
INBOUND ROUTES
Inbound routes can be configured via Web GUI->PBX->Basic/Call Routes->Inbound Routes.

Click on "Create New Inbound Rule" to add a new inbound route.
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
Click on "Blacklist" to configure blacklist for all inbound routes.

Click on
to edit the inbound route.

Click on
to delete the inbound route.
INBOUND RULE CONFIGURATIONS
Table 58: Inbound Rule Configuration Parameters
Trunks
Select the trunk to configure the inbound rule.

All patterns are prefixed with the "_".

Special characters:
X: Any Digit from 0-9.
Z: Any Digit from 1-9.
N: Any Digit from 2-9.
".": Wildcard. Match one or more characters.
"!": Wildcard. Match zero or more characters immediately.
DID Pattern
Example: [12345-9] - Any digit from 1 to 9.

The pattern can be composed of two parts, divided by a ‘/’ character.
The first part is used to specify the dialed number the second part is
used to specify the caller ID and it is optional, if set it means only the
extension with the specific caller ID is allowed to call in or call out.
For example, patter '_2XXX/1234' means the only extension with the
caller ID '1234' is allowed to use this rule.
Privilege Level
Configure the privilege level for this inbound route.
Select the default destination for the inbound call.
Default Destination
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
Extension

Voicemail

Conference Room

Queue

Ring Group

Paging/Intercom

Voicemail Group

Fax

DISA

IVR

Dial By Name

External Number

By DID
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When "By DID" is used, the UCM6100 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 Trunk Name
Configure the number of digits to be stripped from the beginning of the
DID. This option shows up only when "By DID" is selected.
This option shows up only when “By DID” is selected. If enabled, the trunk
name will be prepended to the display name.
This option shows up only when "By DID" is selected. If enabled, the
Dial Trunk
external users dialing in to the trunk via this inbound route can dial
outbound call using the UCM6100’s trunk.
This option shows up only when "By DID" is selected. This controls the
destination that can be reached by the external caller via the inbound
route. The DID destination are:
DID Destination
Alert-Info

Extension

Conference

Call Queue

Ring Group

Paging/Intercom Groups

IVR

Voicemail Groups

Fax Extension

Dial By Name

All
When present in an INVITE request, the Alert-Info header field specifies
an alternative ring tone to the UAS.
Time Condition
Time Conditions
Destination
Select the time condition for the inbound rule.
Select the destination for the inbound call during the specified time
condition.
BLACKLIST CONFIGURATIONS
In the UCM6100, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature and
manage the Blacklist by clicking on "Blacklist".
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
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 remove a number from the Blacklist, select the number in "Blacklist list" and click on
to add to the list.
.
Figure 89: Blacklist Configuration Parameters

To add blacklist number in batch, click on
format is as below.
to upload blacklist file in csv format. The supported csv
Figure 90: 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 UCM6100 supports conference bridge allowing multiple bridges used at the same time:

UCM6102/6104 supports up to 3 conference bridges allowing up to 25 simultaneous PSTN or IP
participants.

The
UCM6108/6116 supports up to 6 conference bridges allowing up to 32 simultaneous PSTN or IP
participants.
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 59: 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
Admin Password
"Enable Caller Menu" is enabled) as well as invite other parties to join the
conference by dialing "0" (permission required from the invited party) or
"1" (permission not required from the invited party) during the conference
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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
Record Conference
automatically in a .wav format file. All the recording files will be displayed
and can be downloaded in the conference web page. The default setting
is "No".
If enabled, if there are users joining or leaving the conference, voice
prompt or notification tone won't be played. The default setting is "No".
Quiet Mode
Note:
"Quiet Mode" and "Announce Callers" cannot be enabled at the same
time.
If enabled, the participants will not hear each other until the conference
administrator joins the conference. The default setting is "No".
Wait For Admin
Note:
If "Quiet Mode" is enabled, the voice prompt for "Wait For Admin" will not
be announced.
If enabled, users could press 0 to invite other users (with the users'
permission) or press 1 to invite other users (without the user's permission)
to join the conference. The default setting is "No".
Enable User Invite
Note:
Conference administrator can always invite other users without enabling
this option.
If enabled, the caller will be announced to all conference participants
when there the caller joins the conference. The default setting is "No".
Announce Callers
Note:
"Quiet Mode" and "Announce Callers" cannot be enabled at the same
time.
Public Mode
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If enabled, no authentication will be required when joining the conference
call. The default setting is "Yes".
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Play Hold Music
If enabled, the UCM6100 will play Hold music when there is only one user
in the conference. The default setting is "No".
Select the music on hold class to be played in conference call. Music On
Music On Hold
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.
INVITE OTHER PARTIES TO JOIN CONFERENCE
When using the UCM6100 conference bridge, there are two ways to invite other parties to join the
conference.

Invite from Web GUI.
For each conference bridge in UCM6100 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 91: Conference Invitation From Web GUI

Invite by dialing 0 or 1 during conference call.
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A conference participant can invite other parties to the conference by dialing from the phone during the
conference call. Please make sure option "Enable User Invite" is turned on for the conference bridge first.
Enter 0 or 1 during the conference call. Follow the voice prompt to input the number of the party you would
like to invite. A call will be sent to this number to join it into the conference.
0: If 0 is entered to invite other party, once the invited party picks up the invitation call, a permission will be
asked to "accept" or "reject" the invitation before joining the conference.
1: If 1 is entered to invite other party, no permission will be required from the invited party.
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.

Manage the conference call from Web GUI.
Log in UCM6100 web GUI during the conference call, the participants in each conference bridge will
be listed.

1. Click on
to kick a participant from the conference.
2. Click on
to mute the participant.
3. Click on
to lock this conference bridge so that other users cannot join it anymore.
4. Click on
to invite other users into the conference bridge.
Manage the conference call from IVR.
If "Enable Caller Menu" is enabled, conference participant can input * to enter the IVR menu for the
conference. Please see options listed in the table below.
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Table 60: Conference Caller IVR Menu
Conference Administrator IVR Menu
1
Mute/unmute yourself.
2
Lock/unlock the conference bridge.
3
Kick the last joined user from the conference.
4
Decrease the volume of the conference call.
5
Decrease your volume.
6
Increase the volume of the conference call.
7
Increase your volume.
More options.
8

1: List all users currently in the conference call.

2: Kick all non-Administrator participants from the conference call.

3: Mute/Unmute all non-Administrator participants from the conference call.

4: Record the conference call.

8: Exit the caller menu and return to the conference.
Conference User IVR Menu
1
Mute/unmute yourself.
4
Decrease the volume of the conference call.
5
Decrease your volume.
6
Increase the volume of the conference call.
7
Increase your volume.
8
Exit the caller menu and return to the conference.
Note:
When there is participant in the conference, the conference bridge configuration cannot be modified.
RECORD CONFERENCE
The UCM6100 allows users to record the conference call and retrieve the recording from web
GUI->PBX->Call Features->Conference.
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To record the conference call, when the conference bridge is in idle, enable "Record Conference" from the
conference bridge configuration dialog. Save the setting and apply the change. When the conference call
starts, the call will be automatically recorded in .wav format.
The recording files will be listed as below once available. Users could click on
recording or click on
to delete the recording.
to download the
Users could also delete all recording files by clicking on
“Delate All Recording Files”, or delete multiple recording files at once by clicking on “Delete Selected
Recording Files” after selecting the recording files.
Figure 92: Conference Recording
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IVR
CONFIGURE IVR
IVR configurations can be accessed under the UCM6100 Web GUI->PBX->Call Features->IVR. Users
could create, edit, view and delete an IVR.

Click on "Create New IVR" to add a new IVR.

Click on
to edit the IVR configuration.

Click on
to delete the IVR.
Table 61: IVR Configuration Parameters
Basic Settings
Name
Configure the name of the IVR. Letters, digits, _ and - are allowed.
Extension
Enter the extension number for users to access the IVR.
Dial Other Extensions
If enabled, all callers to the IVR can dial other extensions. The default
setting is "No".
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
Permission
setting is "Internal". If the user tries to dial outbound calls after dialing into
the IVR, the UCM6100 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.
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 UCM6100 will consider the entries
complete. The default timeout is 3 seconds.
Response Timeout
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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
Language
shows all the current available voice prompt languages on the UCM6100.
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
Select the event for each key pressing for 0-9, *, Timeout and Invalid. The
Key Press Event:
Press 0
Press 1
Press 2
Press 3
Press 4
Press 5
Press 6
Press 7
Press 8
Press 9
Press *
Timeout
Invalid
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event options are:

Extension

Voicemail

Conference Rooms

Voicemail Group

IVR

Ring Group

Queues

Page Group

Fax

Custom Prompt

Hangup

DISA

Dial By Name

External Number

Callback
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CREATE CUSTOM PROMPT
To record new IVR prompt or upload IVR prompt to be used in IVR, click on “Prompt” next to the “Welcome
Prompt” option and the users will be redirected to Custom Prompt page. Or users could go to Web
GUI->PBX->Internal Options->Custom Prompt page directly.
Figure 93: Click on Prompt to Create IVR Prompt
Once the IVR prompt file is successfully added to the UCM6100, it will be added into the prompt list
options for users to select in different IVR scenarios.
RECORD NEW CUSTOM PROMPT
asic Settings
In the UCM6100 web UI->PBX->Internal Options->Custom Prompt page, click on “Record New Custom
Prompt” and follow the steps below to record new IVR prompt.
Figure 94: Record New Custom Prompt
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
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 UCM6100 to record the IVR prompt.

Click the “Record” button. A request will be sent to the UCM6100. The UCM6100 will then call the
extension for recording the IVR prompt from the phone.

Pick up the call from the extension and start the recording following the voice prompt.

The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play or
delete the recording.
UPLOAD CUSTOM PROMPT
If the user has a pre-recorded IVR prompt file, click on “Upload Custom Prompt” in Web
GUI->PBX->Internal Options->Custom Prompt page to upload the file to the UCM6100. The following
are required for the IVR prompt file to be successfully uploaded and used by the UCM6100:

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 95: Upload Custom Prompt
Click on
to select audio file from local PC and click on
to start uploading. Once uploaded, the file
will appear in the Custom Prompt web page.
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LANGUAGE SETTINGS FOR VOICE PROMPT
The UCM6100 supports multiple languages in web GUI as well as system voice prompt. Currently, there
are 16 languages supported in system voice prompt: English (United States), Arabic, Chinese, Dutch,
English (United Kingdom), French, German, Greek, Hebrew, Italian, Polish, Portuguese, Russian,
Spanish, Swedish and Turkish.
English (United States) and Chinese voice prompts are built in with the UCM6100 already. The other
languages provided by Grandstream can be downloaded and installed from the UCM6100 web GUI
directly. Additionally, users could customize their own voice prompts, package them and upload to the
UCM6100.
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 UCM6100 web GUI, click on
"Check Prompt List" button.
Figure 96: Language Settings for Voice Prompt
A new dialog window of voice prompt package list will be displayed. Users can see the version number
(latest version available V.S. current installed version), package size and options to upgrade or download
the language.
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Figure 97: Voice Prompt Package List
Click on
to download the language to the UCM6100. The installation will be automatically started
once the downloading is finished.
Figure 98: New Voice Prompt Language Added
A new language option will be displayed after successfully installed. Users then could select it to apply in
the UCM6100 system voice prompt or delete it from the UCM6100.
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VOICEMAIL
CONFIGURE VOICEMAIL
If the voicemail is enabled for UCM6100 extensions, the configurations of the voicemail can be globally set
up and managed under Web GUI->PBX->Call Features->Voicemail.
Table 62: Voicemail Settings
Max Greeting
Configure the maximum number of seconds for the voicemail greeting.
The default setting is 60 seconds.
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 duration (in seconds) of a voicemail message.
Messages will be automatically deleted if the duration is shorter than the
Min Message Time. The default setting is 3 seconds. The available
options are:
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.
Announce Message Caller-ID
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If enabled, the caller ID of the user who has left the message will be
announced at the beginning of the voicemail message. The default setting
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is "No".
Announce Message Duration
If enabled, the message duration will be announced at the beginning of
the voicemail message. The default setting is "No".
If enabled, a brief introduction (received time, received from, and etc) of
Play Envelope
each message will be played when accessed from the voicemail
application. The default setting is "Yes".
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 UCM6100 extensions, the users can dial the voicemail access feature code
(by default *98 or *97) to access the extension’s voicemail. The users will be prompt to enter the voicemail
password and then can enter digits from the phone keypad to navigate in the IVR menu for different
options.
Table 63: Voicemail IVR Menu
Main Menu
Sub Menu 1
Sub Menu 2
1 - Send a reply
2 - Call the person who sent this message
3 - Advanced options
3 - Hear the message envelop
4 - Leave a message
* - Return to the main menu
1 - New
messages
5 - Repeat the current message
7 - Delete this message
8 - Forward the message to another user
9 – Save
* - Help
# - Exit
0 - New messages
1 - Old messages
2 - Change
2 - Work messages
folders
3 - Family messages
4 - Friend messages
# - Cancel
3-
1 - Send a reply
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Advanced
2 - Call the person who sent this message
options
3 - Hear the message envelop
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 UCM6100 can be configured to send the voicemail as attachment to Email. Click on "Voicemail Email
Settings" button to configure the Email attributes and content.
Table 64: Voicemail Email Settings
Attach Recordings to E-Mail
If enabled, voicemails will be sent to user's Email address. The default
setting is "Yes".
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
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
\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
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message

${VM_MSGNUM}: The number of messages in the mailbox

${VM_DATE}: The date and time when the message is left
Figure 99: Voicemail Email Settings
Click on "Load Default Settings" button to view the default template as an example.
CONFIGURE VOICEMAIL GROUP
The UCM6100 supports voicemail group and all the extensions added in the group will receive the
voicemail to the group extension. The voicemail group can be configured under Web GUI->PBX->Call
Features->Voicemail Group. Click on "Create New Voicemail Group" to configure the group.
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Figure 100: Voicemail Group
Table 65: Voicemail Group Settings
Extension
Name
Voicemail Password
Email Address
Enter the Voicemail Group Extension. The voicemail messages left to this
extension will be forwarded to all the voicemail group members.
Configure the Name to identify the voicemail group. Letters, digits, _ and are allowed.
Configure the voicemail password for the users to check voicemail
messages.
Configure the Email address for the voicemail group extension.
Select available mailboxes from the left list and add them to the right list.
Voicemail Group Mailboxes
The extensions need to have voicemail enabled to be listed in available
mailboxes list.
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RING GROUP
The UCM6100 supports ring group feature with different ring strategies applied to the ring group members.
This section describes the ring group configuration on the UCM6100.
CONFIGURE RING GROUP
Ring group settings can be accessed via Web GUI->PBX->Call Features->Ring Group.
Figure 101: Ring Group

Click on “Create New Ring Group” to add ring group.

Click on
to edit the ring group. The following table shows the ring group configuration parameters.

Click on
to delete the ring group.
Table 66: Ring Group Parameters
Ring Group Name
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 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
Ring Strategy
stop ringing.

Ring in order.
Ring the members with the order configured in ring group list. If the
first member doesn’t answer the call, it will stop ringing the first
member and start ringing the second member.
Configure the permission level for Ring Group. This permission will take
Permission
effect when an incoming call is going to be directed to an external number
configured as “Default Destination”. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
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the highest level.
This option is to set a custom prompt for a ring group to announce to
Custom Prompt
caller. Click on ‘Prompt’, it will direct to the page PBX->Internal
Options->Custom Prompt, where users could record new prompt or
upload prompt files.
Configure the number of seconds to ring each member. If set to 0, it will
keep ringing. The default setting is 30 seconds.
Ring Timeout on Each Member
Note:
The actual ring timeout might be overridden by users if the phone has ring
timeout settings as well.
If enabled, calls on this ring group will be automatically recorded. The
Auto Record
default setting is No. The recording files can be accessed from web
GUI->CDR->Recording Files.
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.
Figure 102: Ring Group Configuration
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PAGING AND INTERCOM GROUP
The UCM6100 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 103: Paging/Intercom Group
Table 67: Paging/Intercom Group Configuration Parameters
Name
Configure paging/intercom group name.
Extension
Configure the paging/intercom group extension.
Type
Select "2-way Intercom" or "1-way Page".
This option is to set a custom prompt for a paging/intercom group to
Custom Prompt
announce to caller. Click on ‘Prompt’, it will direct to the page
PBX->Internal Options->Custom Prompt, where users could record
new prompt or upload prompt files.
Page/Intercom Group
Select available users from the left side to the paging/intercom group
Members
member list on the right.
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
Click on
to edit the paging/intercom group.

Click on
to delete the paging/intercom group.

Click on "Paging/Intercom Group Settings" to edit Alert-Info Header. This header will be included in the
SIP INVITE message sent to the callee in paging/intercom call.
Figure 104: Page/Intercom Group Settings

The UCM6100 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 UCM6100 supports call queue by using static agents or dynamic agents. 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.
Figure 105: Call Queue

Click on "Create New Queue" to add call queue.

Click on
to edit the call queue. The call queue configuration parameters are listed in the table
below.
Table 68: Call Queue Configuration Parameters
Extension
Configure the call queue extension.
Name
Configure the call queue name to identify the call queue.
Select the strategy for the call queue.

Ring All
Ring all available Agents simultaneously until one answers.
Strategy

Linear
Ring agents in the specified order.

Least Recent
Ring the agent who has been called the least recently.
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
Fewest Calls
Ring the agent with the fewest completed calls.

Random
Ring a random agent.

Round Robin
Ring the agents in Round Robin scheduling with memory.
The default setting is "Ring All".
Select the Music On Hold class for the call queue.
Music On Hold
Note:
Music On Hold classes can be managed from Web GUI-> PBX->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 invalid.
Leave When Empty

No
Never disconnect the callers from the queue when the queue is
empty.

Strict
Callers will be disconnected from the queue if all agents are paused,
invalid or unavailable.
Configure whether the callers can dial into a call queue if the queue has
no agent. The default setting is "No".

Yes
Callers can always dial into a call queue.
Dial in Empty Queue

No
Callers cannot dial into a queue if all agents are paused or invalid.

Strict
Callers cannot dial into a queue if the agents are paused, invalid or
unavailable.
Configure the permission level for Call Queue. This permission will take
effect when an incoming call is going to be directed to an external number
Permission
configured as “Default Destination”. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level.
Dynamic Login Password
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log in. The default setting is disabled.
Configure the number of seconds an agent will ring before the call goes to
Ring Time Out
the next agent. The default setting is 15 seconds.
Configure the number of seconds before a new call can ring the queue
Wrapup Time
after the last call on the agent is completed. If set to 0, there will be no
delay between calls to the queue. The default setting is 15 seconds.
Configure the maximum number of calls to be queued at once. This
number does not include calls that have been connected with agents. It
Max Queue Length
only includes calls not connected yet. The default setting is 0, which
means unlimited. When the maximum value is reached, the caller will be
treated with busy tone followed by the next calling rule after attempting to
enter the queue.
If enabled, the UCM6100 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 enabled, the calls on the call queue will be automatically recorded. The
Auto Record
recording files can be accessed in Queue Recordings under web
GUI->PBX->Call Features->Call Queue.
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”.
Failover Destination
Configure the call destination for the call to be routed to if no agent in this
call queue answers the call.
Select the available users to be the static agents in the call queue.
Agents
Choose 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
to delete the call queue.
Extension Postfix. Once configured, users could log in the call queue as dynamic agent.
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Figure 106: Agent Login Settings
For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout
Extension Postfix is **, users could dial 6500* to login to the call queue as dynamic agent and dial
6500** to logout from the call queue. Dynamic agent doesn't need to be listed as static agent and can
log in/log out at any time.

Call queue feature code "Agent Pause" and "Agent Unpause" can be configured under Web
GUI->PBX->Internal Options->Feature Codes. The default feature code is *83 for "Agent Pause"
and *84 for "Agent Unpause".

Queue recordings are shown on the Call Queue page. Click on
.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 UCM6100 extension group feature allows users to assign and categorize extensions in different
groups to better manage the configurations on the UCM6100. 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 107: Edit Extension Group

Click on
to delete the extension group.
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USING EXTENSION GROUPS
Here is an example where the extension group can be used. Go to Web GUI->PBX->Basic/Call
Routes->Outbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and
extension groups will show up for users to select.
Figure 108: Select Extension Group in Outbound Route
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PICKUP GROUPS
The UCM6100 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 109: 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 110: 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 111: Music On Hold Default Class

Click on "Create New MOH Class" to add a new Music On Hold class.
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
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 8 KHz Mono format with size smaller than 5M.

Click on

Select the sound files and click on
next to the sound file to delete it from the selected Music On Hold Class.
to delete all selected music on
hold files.
Note:
Once the MOH file is deleted, there are two ways to recover the music files.

Users could download the MOH file from this link:
http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-moh-opsound-wav-2.03.tar.gz
After downloading and unzip the pack, users could then upload the music files to UCM.

Factory reset could also recover the MOH file on the UCM.
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FAX/T.38
The UCM6100 supports T.30/T.38 Fax and Fax Pass-through. It can convert the received Fax to PDF
format and send it to the configured Email address. Fax/T.38 settings can be accessed via Web
GUI->PBX->Internal Options->FAX/T.38. The list of received Fax files will be displayed in the same web
page for users to view, retrieve and delete.
CONFIGURE FAX/T.38

Click on "Create New Fax Extension". In the popped up window, fill the extension, name and Email
address to send the received Fax to.

Click on "Fax Settings" to configure the Fax parameters.
Table 69: FAX/T.38 Settings
Enable Error Correction Mode
Configure to enable Error Correction Mode (ECM) for the Fax. The default
setting is "Yes".
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 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 sending the Fax to the users.
The template variables are:
Template Variables
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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
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
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 describes how to use the UCM6100 to receive Fax from PSTN line on the Fax
machine connected to the UCM6100 FXS port.
1. Connect Fax machine to the UCM6100 FXS port.
2. Connect PSTN line to the UCM6100 FXO port.
3. Go to web GUI->PBX->Analog Trunks page.
4. Create or edit the analog trunk for Fax as below.
Fax Mode: Make sure "Fax Mode" option is set to "None".
Figure 112: Configure Analog Trunk without Fax Detection
5. Go to UCM6100 web GUI->PBX->Basic/Call Routes->Extensions page.
6. Create or edit the extension for FXS port.
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
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 113: Configure Extension for Fax Machine
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 114: Configure Inbound Rule for Fax
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Now the Fax configuration is done. When there is an incoming Fax calling to the PSTN number for the
FXO port, it will send the Fax to the Fax machine.
SAMPLE CONFIGURATION FOR FAX-TO-EMAIL
The following instructions describes a sample configuration on how to use Fax-to-Email feature on the
UCM6100.
1. Connect PSTN line to the UCM6100 FXO port.
2. Go to UCM6100 web GUI->Internal Options->Fax/T.38 page. Create a new Fax extension.
Figure 115: Create Fax Extension
3. Go to UCM6100 web GUI->Basic/Call Routes->Analog Trunks page. Create a new analog trunk.
Please make sure "Fax Detection" is set to "No".
4. Go to UCM6100 web GUI->Basic/Call Routes->Inbound Routes page. Create a new inbound route
and set the default destination to the Fax extension.
Figure 116: Inbound Route to Fax Extension
5. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number
will be converted to PDF file and sent to the Email address Faxtest@ucm6100mycompany.com as
attachment.
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ASTERISK MANAGER INTERFACE (RESTRICTED ACCESS)
Starting from firmware 1.0.8.11, the UCM6100 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 UCM6100 web GUI->PBX->Internal Options->AMI. For details
on how to use AMI on UCM6100, please refer to the following AMI guide:
http://www.grandstream.com/products/ucm_series/ucm61xx/documents/ucm6100_ami_guide.pdf
Warning:
Please do not enable AMI on the UCM6100 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 UCM6100 system. Please be cautious when enabling AMI access on the UCM6100 and restrict the
permission granted to the AMI user. By using AMI on UCM6100 you agree you understand and
acknowledge the risks associated with this.
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BUSY CAMP-ON
Starting from firmware 1.0.8.11, the UCM6100 supports busy camp-on/call completion feature that allows
the PBX to camp on a called party and inform the caller as soon as the called party becomes available
given the previous attempted call has failed.
The configuration and instructions on how to use busy camp-on/call completion feature can be found in the
following guide:
http://www.grandstream.com/products/ucm_series/ucm61xx/documents/ucm6100_busy_camp_on_guide.
pdf
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FOLLOW ME
Follow Me is a feature on the UCM6100 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.
To configure follow me:

Click on "Create New Follow Me" and then select an extension to be configured with Follow Me.
Figure 117: Create Follow Me

Click on “Next” to continue editing Follow Me configuration.
Figure 118: Edit Follow Me

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.
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
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 70: Follow Me Settings
Enable
Music On Hold Class
Configure to enable or disable Follow Me for this user.
Configure the Music On Hold class that the caller would hear while
tracking the user.
The added numbers are listed here. Click on
Follow Me Numbers
order. Click on
numbers.
to arrange the
to delete the number. Click on
to add new
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.
Click on “Follow Me Options” to enable or disable the options listed in the following table.
Table 71: 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 Status
If enabled, the PBX will playback the unreachable status message to the
Message
caller if the callee cannot be reached.
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ONE-KEY DIAL
The UCM6100 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 UCM6100.
To enable One-Key Dial, on the UCM6100 web GUI, go to page PBX->Call Features->One-Key Dial.
Figure 119: 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 120: One-Key Dial Destinations
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DISA
The UCM6100 supports DISA to be used in IVR or inbound route. Before using it, create new DISA under
web GUI->Call Features->DISA.

Click on "Create New IVR" to add a new DISA.

Click on
to edit the DISA configuration.

Click on
to delete the DISA.
Figure 121: Create New DISA
Table 72: 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.
Configure the permission level for DISA. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
Permission
the highest level. The default setting is "Internal". If the user tries to dial
outbound calls after dialing into the DISA, the UCM6100 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
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privilege level, the call will be allowed to go through.
Configure the maximum amount of time the UCM6100 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 UCM6100 hangup
Allow Hangup
feature code (by default it's *0) to disconnect the call or hang up directly. A
new dial tone will be heard shortly for the user to make a new call. The
default setting is "No".
Once successfully created, users can configure the inbound route destination as "DISA" or IVR key event
as "DISA". When dialing into DISA, users will be prompted with password first. After entering the correct
password, a second dial tone will be heard for the users to dial out.
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CALLBACK FEATURE
Callback is mainly designed for users who often use their mobile phones to make long distance or
international calls which may have high service charges. The callback feature provides an economic
solution for reduce the cost from this.
The callback feature works as follows:
1. Configure a new callback on the UCM6100.
2. On the UCM6100, 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 UCM6100 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 UCM6100 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 UCM6100 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 UCM6100, go to web GUI->PBX->Call Features->Callback page and click
on
. Configuration parameters are listed in the following table.
Table 73: Callback Configuration Parameters
Name
Configure a name to identify the Callback.
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
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Configure the number of seconds to be delayed before calling back the
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user.
Configure the destination which the callback will direct the caller to. Two
destinations are available:
Destination

IVR

DISA
The caller can then enter the desired number to dial out via UCM6100
trunk.
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BLF AND EVENT LIST
BLF
The UCM6100 supports BLF monitoring for extensions, ring group, call queue, conference room and
parking lot. For example, on the user's phone, configure the parking lot number 701 as the BLF monitored
number. When there is a parked call on 701, the LED for this BLF key will light up in red, meaning a call is
parked against this parking lot. Pressing this BLF key can pick up the call from this parking lot.
Note:
On the Grandstream GXP series phones, the MPK supports "Call Park" mode, which can be used to park
the call by configuring the MPK number as call park feature code (e.g., 700). MPK "Call Park" mode can
also be used to monitor and pickup parked call if the MPK number is configured as parking lot (e.g., 701).
EVENT LIST
Besides BLF, users can also configure the phones to monitor event list. In this way, both local extensions
on the same UCM6100 and 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.

Click on
to edit the event list configuration.

Click on
to delete the event list.
Table 74: Event List Settings
Configure the name of this event list (for example, office_event_list).
URI
Please note the URI name cannot be the same as the extension name on
the UCM6100. The valid characters are letters, digits, _ and -.
Local Extensions
Select the available extensions/Extension Groups listed on the local
UCM6100 to be monitored in the event list.
If LDAP sync is enabled between the UCM6100 and the peer UCM6100,
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
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Figure 122: Create New Event List
Remote extension monitoring works on the UCM6100 via event list BLF, among Peer SIP trunks or
Register SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the
UCM6100 first before using remote BLF feature. Please note the SIP end points need support event list
BLF in order to monitor remote extensions.
When an event list is created on the UCM6100 and remote extensions are added to the list, the UCM6100
will send out SIP SUBSCIRBE to the remote UCM6100 to obtain the remote extension status. When the
SIP end points registers and subscribes to the local UCM6100 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 UCM6100 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 UCM6100 to connect to the local UCM6100. Additional information such as port number,
LDAP outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM6100 and remote
UCM6100 need enable LDAP sync option with the same password for successful connection and
synchronization.

Currently LDAP sync feature only works between two UCM6100s.

(Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is
non-UCM6100 PBX. However, it might not work the other way around depending on whether the
non-UCM6100 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 123: Create Dial By Name Group
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1. Group Name
Enter the Group Name. This is to identify the Dial By Name group. The Dial By Name group can be used
as the destination for inbound route and key pressing event for IVR. The group name defined here will
show up in the destination list when configuring IVR and inbound route.
Figure 124: Dial By Name Group In IVR Key Pressing Events
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Figure 125: Dial By Name Group In Inbound Rule
2. Extension
Configure the direct dial extension for the Dial By Name group.
3. Available Extensions/Selected Extensions
Select available extensions from the left side to the right side as the directory for the Dial By Name group.
Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group.
The extensions here must have a valid first name and last name configured under web
GUI->PBX->Basic/Call Routes->Extensions in order to be searchable in Dial By Name directory through
IVR. By specifying the extensions here, the administrators can make sure unscreened calls will not reach
the company employee if he/she doesn't want to receive them directly.
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Figure 126: 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 UCM6100 are displayed in web UI->Status->Active Calls page. Users can monitor
the status, hang up the call as well as barge in the active calls in real time manner.
ACTIVE CALLS STATUS
To view the status of active calls, navigate to web GUI->Status->Active Calls. The following figure shows
extension 1000 is calling 1001. 1001 is ringing.
Figure 127: Status->PBX Status->Active Calls - Ringing
The following figure shows the call between 1000 and 1001 is established.
Figure 128: Status->PBX Status->Active Calls – Call Established
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In active call web page, click on
to refresh the active call status.
HANG UP ACTIVE CALLS
To hang up an active call, click on
icon in the active call dialog. Users can also click on
to hang up all active calls.
CALL MONITOR
During an active call, click on icon
and the monitor dialog will pop up.
Figure 129: Configure to Monitor an Active Call
In the “Monitor” dialog, configure the following to monitor an active call:
1. Enter an available extension for “Monitor’s Extension” which will be used to monitor the active call.
2. “Monitored Extension” must be one of the parties in the active call to be monitored.
3. Select spy mode. There are three options in “Spy Mode”.

Listen
In “Listen” mode, the extension monitoring the call can hear both parties in the active call but the
audio of the user on this extension will not be heard by either party in the monitored active call.

Whisper
In “Whisper” mode, the extension monitoring the call can hear both parties in the active call. The
user on this extension can only talk to the selected monitored extension and he/she will not be
heard by the other party in the active call. This can be usually used to supervise calls.

Barge
In “Barge” mode, the extension monitoring the call can talk to both parties in the active call. The
call will be established similar to three-way conference.
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4. Enable or disable “Require Confirmation” option. If enabled, the confirmation of the invited monitor’s
extension is required before the active call can be monitored. This option can be used to avoid adding
participant who has auto-answer configured or call forwarded to voicemail.
5. Click on “Add”. An INVITE will be sent to the monitor’s extension. The monitor can answer the call and
start monitoring. If “Require Confirmation” is enabled, the user will be asked to confirm to monitor the
call.
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CALL FEATURES
The UCM6100 supports call recording, transfer, call forward, call park and other call features via feature
code. This section lists all the feature codes in the UCM6100 and describes how to use the call features.
FEATURE CODES
Table 75: UCM6100 Feature Codes
Feature Maps

Default code: #1.

Enter the code during active call. After hearing "Transfer", you
will hear dial tone. Enter the number to transfer to. Then the
user will be disconnected and transfer is completed.
Blind Transfer

Options
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.

Default code: *2.

Enter the code during active call. After hearing "Transfer", you
will hear the dial tone. Enter the number to transfer to and the
user will be connected to this number. Hang up the call to
Attended Transfer
complete the attended transfer.

Options
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.
Disconnect

Default code: *0.

Enter the code during active call. It will disconnect the call.

Options
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.
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
Default code: #72.

Enter the code during active call to park the call.

Options
Call Park
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.

Default code: *3.

Enter the code followed by # or SEND to start recording the
audio call and the UCM6100 will mix the streams natively on
the fly as the call is in progress.
Audio Mix Record

Options
Disable
Allow Caller: Enable the feature code on caller side only.
Allow Callee: Enable the feature code on callee side only.
Allow Both: Enable the feature code on both caller and callee.
DND/Call Forward
Do Not Disturb (DND) Activate

Default code: *77.
Do Not Disturb (DND) Deactivate

Default code: *78.

Default Code: *90.

Enter the code and follow the voice prompt. Or enter the code
Call Forward Busy Activate
followed by the extension to forward the call.
Call Forward Busy Deactivate
Call Forward No Answer Activate

Default Code: *91.

Default Code: *92.

Enter the code and follow the voice prompt. Or enter the code
followed by the extension to forward the call.
Call Forward No Answer
Deactivate
Call Forward Unconditional
Activate
Call Forward Unconditional
Deactivate

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.
Feature Misc
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Feature Code Digits Timeout

Default Setting: 1000.

Configure the maximum interval (in milliseconds) between the
digits input to activate the feature code.
Call Park

Default Extension: 700.

During an active call, initiate blind transfer and then enter this
code to park the call.
Parked Lots

Default Extension: 701-720.

These are the extensions where the calls will be parked, i.e.,
parking lots that the parked calls can be retrieved.
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.
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Blacklist Remove

Default Code: *41.

To remove a number from current blacklist for inbound route,
dial *41 and follow the voice prompt to remove the number.
Call Pickup on Ringing

Default Code: **.

To pick up a call for any extension xxxx, enter the code followed
by the extension number xxxx.
Pickup Extension

Default Code: *8.

This code is for the pickup group which can be assigned for
each extension on the extension configuration page.

If there is an incoming call to an extension, the other
extensions within the same pickup group can dial *8 directly to
pick up the call.

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.
CALL RECORDING
The UCM6100 allows users to record audio during the call. If "Auto Record" is turned on for an extension,
ring group, call queue or trunk, the call will be automatically recorded when there is established call with it.
Otherwise, please follow the instructions below to manually record the call.
1. Make sure the feature code for "Audio Mix Record" is configured and enabled.
2. After establishing the call, enter the "Audio Mix Record" feature code (by default it's *3) followed by #
or SEND to start recording.
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3. To stop the recording, enter the "Audio Mix Record" feature code (by default it's *3) followed by # or
SEND again. Or the recording will be stopped once the call hangs up.
4. The recording file can be retrieved under Web GUI->Status->CDR. Click on
or click on
to play the recording
to download the recording file.
Figure 130: Download Recording File from CDR Page
The above recorded call's recording files are also listed under the UCM6100 web GUI->CDR->Recording
Files.
Figure 131: Download Recording File from Recording Files Page
CALL PARK
The UCM6100 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.
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
Feature Misc->Call Park (Default code 700)
During an active call, initiate blind transfer (default code #1) and then dial 700 to park the call. Parking
lot number (default range 701 to 720) will be announced after parking the call.
RETRIEVE 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.
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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 UCM6100, including general configurations, jitter buffer, RTP
settings, ports config and STUN monitor. The options can be accessed via Web GUI->PBX->Internal
Options.
INTERNAL OPTIONS/GENERAL
Table 76: 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 presses 0
Operator Extension
to 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.
If enabled, users will hear voice prompt before recording is started or
Record Prompt
stopped. For example, before recording, the UCM6100 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 UCM6100. The default setting is enabled.
Enforce Strong Passwords
Strong Password Rules:

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.

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.
If
set
to
"Yes",
users
could
disable
the
extension
range
pre-configured/configured on the UCM6100. The default setting is "No".
Disable Extension Range
Note:
It is recommended to keep the system assignment to avoid inappropriate
usage and unnecessary issues.
The default extension range assignment is:

User Extensions: 1000-6299
User Extensions is referring to the extensions created under web
UI->PBX->Basic/Call Routes->Extensions page.

Pick Extensions: 4000-4999
This refers to the extensions that can be manually picked from end
device when being provisioned by the UCM6100. There are two
related options in zero config page->Auto Provision Settings, "Pick
Extension Ranges
Extension Segment" and "Enable Pick Extension". If "Enable Pick
Extension" under zero config settings is selected, the extension list
defined in "Pick Extension Segment" will be sent out to the device
after receiving the device's request. This "Pick Extension Segment"
should be a subset of the "Pick Extensions" range here. This feature
is for the GXP series phones that support selecting extension to be
provisioned via phone's LCD.

Auto Provision Extensions: 5000-6299
This sets the range for "Zero Config Extension Segment" which is the
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extensions can be assigned on the UCM6100 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

Fax Extensions: 7200-8200
INTERNAL OPTIONS/JITTER BUFFER
Table 77: Internal Options/Jitter Buffer
SIP Jitter Buffer
Enable Jitter Buffer
Select to enable jitter buffer on the sending side of the SIP channel. The
default setting is "No".
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".

Implementation
Fixed
The size is always equal to the value of "Max Jitter Buffer".

Adaptive
The size is adjusted automatically and the maximum value equals to
the value of "Max Jitter Buffer".
INTERNAL OPTIONS/RTP SETTINGS
Table 78: Internal Options/RTP Settings
RTP Start
Configure the RTP port starting number. The default setting is 10000.
RTP End
Configure the RTP port ending address. The default setting is 20000.
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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".
INTERNAL OPTIONS/PORTS CONFIG
The analog hardware (FXS port and FXO port) on the UCM6100 will be listed in this page. Click on
to
edit signaling preference for FXS port or configure ACIM settings for FXO port.
Select "Loop Start" or "Kewl Start" for each FXS port. And then click on "Update" to save the change.
Figure 132: 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 UCM6100 to automatically detect the ACIM value. The
detecting value will be automatically filled into the settings.
Figure 133: FXO Ports ACIM Settings
Table 79: Internal Options/Ports Config
Select country to set the default tones for dial tone, busy tone, ring tone
Tone Region
and etc to be sent from the FXS port. The default setting is "United States
of America (USA)".
Advanced Settings
FXO Opermode
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Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum
Operational Loop Current, and AC Impedance as predefined for your
country's analog line characteristics. The default setting is "United States
of America (USA)".
Select country to set the On Hook Speed, Ringer Impedance, Ringer
Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum
FXS Opermode
Operational Loop Current, and AC Impedance as predefined for your
country's analog line characteristics. The default setting is "United States
of America (USA)".
Configure to enable or disable override Two-Wire Impedance Synthesis
(TISS). The default setting is No.
FXS TISS Override
If enabled, users can select the impedance value for Two-Wire
Impedance Synthesis (TISS) override. The default setting is 600Ω.
Select the codec to be used for analog lines. North American users
should choose PCMU. All other countries, unless already known, should
PCMA Override
be assumed to be PCMA. The default setting is PCMU.
Note:
This option requires system reboot to take effect.
Configure whether normal ringing voltage (40V) or maximum ringing
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
FXS MWI Mode
default setting is "FSK".

FSK: Frequency Shift Key Indicator

NEON: Light Neon Bulb Indicator.
INTERNAL OPTIONS/STUN MONITOR
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Table 80: Internal Options/STUN Monitor
Configures the IP address or URL of the STUN server to query. If not
specified, STUN is disabled. The default setting is stun.ipvideotalk.com.
STUN Server
Valid format:
[(hostname | IP-address) [':' port]
The default port number is 3478 if not specified.
STUN Refresh
Configure the number of seconds between STUN Refreshes. The default
setting is 30 seconds.
INTERNAL OPTIONS/PAYLOAD
The UCM6100 payload type for audio codecs and video codes can be configured here.
Table 81: 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.
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.
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IAX SETTINGS
The UCM6100 IAX global settings can be accessed via Web GUI->PBX->IAX Settings.
IAX SETTINGS/GENERAL
Table 82: 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
Music On Hold Interpret
when being put on hold. This music class is only effective if this channel
has no music class configured and the bridged channel putting the call on
hold has no "Music On Hold Suggest" setting.
Music On Hold Suggest
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 83: IAX Settings/Registration
IAX Registration Options
Min Reg Expire
Max Reg Expire
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.
IAX Thread Count
Configure the number of IAX helper threads. The default setting is 10.
IAX Max Thread Count
Configure the maximum number of IAX threads allowed. The default
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setting is 100.
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 84: IAX Settings/Static Defense
Call Token Optional
Enter a single IP address (e.g., 11.11.11.11) or a range of IP addresses
(11.11.11.11/22.22.22.22) for which call token validation is not required.
Max Call Numbers
Configure the maximum number of calls allowed for a single IP address.
Max Unvalidated Call Numbers
Configure the maximum number of unvalidated calls for all IP addresses.
Call Number Limits
Configure to limit the number of calls for a give IP address of IP range.
IP or IP Range
Firmware Version 1.0.8.12
Enter the IP address (11.11.11.11) or a range of IP addresses
(11.11.11.11/22.22.22.22) to be considered for call number limits.
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SIP SETTINGS
The UCM6100 SIP global settings can be accessed via Web GUI->PBX->SIP Settings.
SIP SETTINGS/GENERAL
Table 85: SIP Settings/General
Realm For Digest
Authentication
Bind UDP Port
Bind IP Address
Configure the host name or domain name for the UCM6100. 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 UCM6100 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 UCM6100 address to dial into
the UCM6100.
Overlap Dialing
Allow Transfer
Select to enable overlap dialing support. The default setting is "No".
If set to "No", all transfers initiated by the endpoint in the UCM6100 will be
disabled (unless enabled in peers or users). The default setting is "Yes".
Enable DNS SRV Lookups on
Select to enables DNS SRV lookups on outbound calls from the
Outbound Calls
UCM6100. The default setting is "Yes".
When sending MWI NOTIFY requests, this value will be used in the
MWI From
"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 Domain Support
Configure the domain for the UCM6100. Incoming INVITE and REFER
messages can be matched against a list of "allowed" domains, each of
which can direct the call to a specific context if desired. By default, all
Domain
domains are accepted and sent to the default context or the context
associated with the user/peer placing the call. Register to non-local
domains will be automatically denied if a domain list is configured. Up to
10 domains can be added.
From Domain
Firmware Version 1.0.8.12
Configure the domain in the "From:" header of the SIP message. It may
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be required by some providers for authentication.
Auto Domain
Allow External Domains
If enabled, the UCM6100 will add local host name and local IP to domain
list. The default setting is "No".
If enabled, requests for external domains that are not served by the
UCM6100 will be allowed. The default setting is "Yes".
SIP SETTINGS/MISC
Table 86: 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 UCM6100 gives
Register Attempts
up. The default setting is 0, which means the UCM6100 will keep trying
until the server side accepts the registration request.
Video
Max Bit Rate (kb/s)
Support SIP Video
Generate Manager Events
Configure the maximum bit rate (in kb/s) for video calls. The default
setting is 384.
Select to enable video support in SIP calls. The default setting is "Yes".
If enabled, the UCM6100 will generate manager events when SIP UA
performs events (e.g. Hold). The default setting is "No".
If enabled, when rejecting an incoming INVITE or REGISTER request, the
UCM6100 will always reject with "401 Unauthorized" instead of notifying
Reject Non-Matching INVITE
the requester whether there is a matching user or peer for the request.
This reduces the ability of an attacker to scan for valid SIP usernames.
The default setting is "No".
SIP SETTINGS/SESSION TIMER
Table 87: 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.
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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
Table 88: SIP Settings/TCP and TLS
TCP Enable
Configure to allow incoming TCP connections with the UCM6100. 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.
TLS Enable
Configure to allow incoming TLS connections with the UCM6100. 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
self-signed certificate, including server's public key. This file will be
TLS Self-Signed CA
renames as "TLS.ca" automatically.
Note:
The size of the uploaded ca file must be under 2MB.
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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
TLS CA Cert
accessed servers.
Note:
The size of the uploaded CA certificate file must be under 2MB.
TLS CA List
Display a list of files under the CA Cert directory.
SIP SETTINGS/NAT
Table 89: SIP Settings/NAT
Configure a static address and port (optional) that will be used in
External IP Address
outbound SIP messages if the UCM6100 is behind NAT. If it's a
hostname, it will only be looked up once.
Specify an external host name, which is similar to External Address
External Host
except the host name will be looked up periodically based on the
"External Refresh" interval.
External Refresh
External TCP Port
External TLS Port
Configure the refresh interval for the external host (if used) The default
setting is 10.
Configure the externally mapped TCP port when the UCM6100 is behind
a static NAT or PAT.
Configures the externally mapped TLS port when UCM6100 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
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SIP SETTINGS/TOS
Table 90: 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
Configure the maximum duration (in seconds) of incoming registration
Time
and subscription allowed by the UCM6100. The default setting is 3600.
Min Registration/Subscription
Configure the minimum duration (in seconds) of incoming registration and
Time
subscription allowed by the UCM6100. The default setting is 60.
Configure the Music On Hold class for the channel when being put on
Music On Hold Interpret
hold. This is used when the Music On Hold class is not set on the channel
and the peer channel placing the call on hold doesn't have "Music On
Hold Suggest".
Music On Hold Suggest
Enable Relaxed DTMF
Configure the Music On Hold class to suggest to the peer channel when
placing the peer on hold.
Select to enable relaxed DTMF handling. The default setting is "No".
Select DTMF mode to send DTMF. The default setting is RFC2833. If
"Info" is selected, SIP INFO message will be used. If "Inband" is selected,
DTMF Mode
64-kbit codec PCMU and PCMA are required. When "Auto" is selected,
"RFC2833" will be used if offered, otherwise "Inband" will be used. The
default setting is "RFC2833".
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
Firmware Version 1.0.8.12
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".
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Configure whether the UCM6100 should generate inband ringing or not.
The default setting is "Never".

Yes: The UCM6100 will send 180 Ringing followed by 183 Session
Progress and in-band audio.
Generate In-Band Ringing

No: The UCM6100 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 UCM6100 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 UCM6100.
Send Compact SIP Headers
If enabled, compact SIP headers will be sent. The default setting is "No".
Add "user=phone" to URI
Firmware Version 1.0.8.12
If enabled, "user=phone" will be added to URI that contains a valid phone
number. The default setting is "No".
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STATUS AND REPORTING
PBX STATUS
The UCM6100 monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces and
Parking lot. It presents administrators the real time status in different sections under web
GUI->Status->PBX Status.
Figure 134: Status->PBX Status
TRUNKS
Users could see all the configured trunk status in this section.
Figure 135: Trunk Status
Table 91: Trunk Status
Display trunk status.

Status
Analog trunk status:
Available
Busy
Unavailable
Unknown Error
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
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
Trunks
Display trunk name
Display trunk Type:
Type

Analog

SIP

IAX
Username
Display username for this trunk.
Port/Hostname/IP
Display Port for analog trunk, or Hostname/IP for VoIP (SIP/IAX) trunk.
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
Users could see all the configured extension status in this section.
Figure 136: Extension Status
Table 92: Extension Status
Status
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Display extension number (including feature code). The color indicator has the
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following definitions.

Green: Free

Blue: Ringing

Yellow: In Use

Grey: Unavailable
Extension
Display the extension number.
Name/Label
First name and last name of the extension.
Display message status for the extension.
Example: 2/4/1
Message
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

Ring Groups

Voicemail Groups
Other operations are also available in extension status section:

Click on "Extensions", the web page will redirect to extension configuration page which can also be
accessed via web GUI->PBX->Basic/Call Routes->Extensions.

Click on

Click on one of the tabs
to refresh the extension status.
to display
the corresponding extensions accordingly.

Click on [ + ] to expand the status detail table.

Click on [ - ] to hide the status detail table.
QUEUES
Users could see all the configured call queue status in this section. The following figure shows the call
queue 6500 being in used.
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Figure 137: 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 93: Agent Status
The agent is available/idle.
The agent is ringing.
The agent is talking/busy.
The agent has been logged out.
On the UCM6100, 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.
CONFERENCE ROOMS
Users could see all the conference room status in this section. It shows all the configured conference
rooms, current users, call duration for each user and conference call.
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Figure 138: 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/port connection status on the UCM6100. The following example shows the
interface status for UCM6116 with USB, SD card, LAN port and FXS1 connected.
Figure 139: UCM6116 Interfaces Status
Table 94: Interface Status Indicators
USB connected.
USB disconnected.
SD Card connected.
SD Card disconnected.
LAN/WAN connected.
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LAN/WAN not configured.
LAN/WAN disconnected.
FXS/FXO connected.
FXS/FXO waiting.
FXS/FXO busy.
FXS/FXO not configured.
FXS/FXO disconnected.
Other operations are also available in interface status section:

Click on "Interfaces Status", the web page will redirect to ports configuration page which can also be
accessed via web GUI->PBX->Internal Options->Ports Config.

Click on

Click on [ + ] to expand the interface details.

Click on [ - ] to hide the interface details.
to refresh the interface status.
PARKING LOT
The UCM6100 supports call park using feature code. When there is call being parked, this section will
display the parking lot status.
Figure 140: Parking Lot Status
Table 95: 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
Timeout
dynamically update this timer from 120 seconds (default) to 0. When the timer
reaches 0, the caller who parks the call will be called back.
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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 UCM6100 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 UCM6100. Please see details in the following table.
Table 96: System Status->General
Status ->System Status -> General
Model
Product model.
Part Number
Product part number.
System Time
Current system time. The current system time is also available on the upper right of
each web page.
Up Time
System up time since the last reboot.
Boot
Boot version.
Core
Core version.
Base
Base version.
Program
Program version. This is the main software release version.
Recovery
Recovery version.
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NETWORK
Under Web GUI->Status->System Status->Network, users could check the network information for the
UCM6100. Please see details in the following table.
Table 97: 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
Voicemail, recording files, IVR file, Music on Hold files and etc.

USB disk
USB disk will display if connected.

SD Card
SD Card will display if connected.
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Figure 141: System Status->Storage Usage
RESOURCE USAGE
When configuring and managing the UCM6100, 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 142: System Status->Resource Usage
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SYSTEM EVENTS
The UCM6100 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 UCM6100 which will have alert and/or Email generated
if occurred:
Disk Usage
Modify Admin Password
Memory Usage
System Reboot
System Update
System Crash
Register SIP Failed
Register SIP Trunk Failed
Restore Config
User Login Success
User Login Failed
SIP Internal Call Failure
SIP Outgoing Call through Trunk Failure
Fail2ban Blocking
SIP Lost Registration
SIP Peer Trunk Status
Click on
to configure the parameters for each event. See examples below.
1. Disk Usage.
Figure 143: System Events->Alert Events Lists: Disk Usage
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
Detect Cycle: The UCM6100 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 UCM6100
system will send the alert.
2. Memory Usage
Figure 144: System Events->Alert Events Lists: Memory Usage

Detect Cycle: The UCM6100 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 UCM6100
system will send the alert.
3. System Reboot
Figure 145: System Events->Alert Events Lists: System Reboot

Detect Cycle: The UCM6100 will check the system reboot based on this cycle. Users can enter
the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
4. System Crash
Figure 146: System Events->Alert Events Lists: System Crash
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
Detect Cycle: The UCM will detect the event at each cycle based on the specified time. Users can
enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.
Click on the switch
to turn on/off the alert and Email notification for the event. Users could
also select the checkbox for each event and then click on button "Alert On", "Alert Off", "Email Notification
On", "Email Notification Off" to control the alert and Email notification configuration.
ALERT LOG
Under Web GUI->Status->System Events->Alert Log, system messages are listed for the alert from
triggered system events. The following picture shows disk usage and user login success alert log. We can
tell the detect cycle for the disk usage is 10 minutes.
Figure 147: System Events->Alert Log
User could also filter out alert logs of a certain event category within a certain time period. Select a
category of event interested in in “Event Name” first. Then select “From Date” and “To Date”. Click on
“View Report” to view the filtered result. The following figure shows an example of filtering out all “Register
SIP Failed” alert logs within the time period from October 1st 00:00, 2014 to November 3rd 23:59, 2014.
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Figure 148: Filter for Alert Log
ALERT CONTACT
Users could add administrator's Email address under Web GUI->Status->System Events->Alert Contact
to send the alert notification to. Up to 10 Email addresses can be added.
CDR
A Call Detail Record (CDR) is a data record produced by telephone exchange activities or other
telecommunications equipment documenting the details of a phone call that passed through the PBX. The
CDR is composed of the following data fields on the UCM6100.

Start Time. Format: 2013-03-27 16:47:03.

Call From. Format: "John Doe"<6012>.

Call To. Format: 6005.

Answered By. Format: 6005.

Call Time. Format: 0:00:10.

Talk Time. Format: 0:00:10

Status. Format: NO ANSWER, BUSY, ANSWERED, or FAILED.

Options. Voice record playing/downloading/deleting.
Users could filter the call report by specifying the date range and criteria, depending on how the users
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would like to include the logs to the report. Then click on "View Report" button to display the generated
report.
Figure 149: CDR Filter
Table 98: 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.
Caller Number
Enter the caller number to be filtered in the CDR report.
Caller Name
Enter the caller name to be filtered in the CDR report.
From Date
Specify "From" date and time to be filtered for the CDR report. Click on the field and
the calendar will show for users to select the exact date and time.
To Date
Specify "To" date and time to be filtered for the CDR report. Click on the field 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 150: Call Report
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Users could perform the following operations on the call report.

Sort
Click on the header of the column to sort by this category. For example, clicking on "Start Time" will
sort the report according to start time. Clicking on "Start Time" again will reverse the order.

Download Searched Results
Click on “Download Search Result(s)” to export the records filtered out to a .csv file.

Download All Records
Click on “Download All Records” to export all the records to a .csv file.

Delete All
On the bottom of the page, click on "Delete All" button to remove all the call report information.

Play/Download/Delete Recording File (per entry)
If the entry has audio recording file for the call, the three icons on the most right column will be
activated for users to select. In the following picture, the second entry has audio recording file for the
call.
Click on
on
to play the recording file; click on
to download the recording file in .wav format; click
to delete the recording file (the call record entry will not be deleted).
Figure 151: Call Report Entry with Audio Recording File

Automatic Download CDR Records
User could configure the UCM6100 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.
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Figure 152: Automatic Download Settings
To receive CDR record automatically from Email, check “Enable” and select a time period “By Day” “By
Week” or “By Month” for the automatic download period. Make sure you have entered an Email
address to receive the CDR records.
DOWNLOADED CDR FILE
The downloaded CDR (.csv file) has different format from the web UI CDR. Here are some descriptions.

Call From, Call To
"Call From": the caller ID.
"Call To": the callee ID.
If "Call From" shows empty, "Call To" shows "s" (see highlight part in the picture below) and the "Source
Channel" contains "DAHDI", this means the call is from FXO/PSTN line. For FXO/PSTN line, we only know
there is an incoming request when there is incoming call but we don't know the number being called. So
we are using "s" to match it where "s" means "start".
Figure 153: Downloaded CDR File Sample - Call To Shows "s"

Context
There are different context values that might show up in the downloaded CDR file. The actual value can
vary case by case. Here are some sample values and their descriptions.
from-internal: internal extension makes outbound calls.
ext-did-XXXXX: inbound calls. It starts with "ext-did", and "XXXXX" content varies case by case, which
also relate to the order when the trunk is created.
ext-local: internal calls between local extensions.

Source Channel, Dest Channel
Sample 1:
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Figure 154: Downloaded CDR File Sample - Source Channel and Dest Channel 1
DAHDI means it is an analog call, FXO or FXS.
For UCM6102, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports.
For UCM6104, DAHDI/(1-4) are FXO ports, and DAHDI(5-6) are FXS ports.
For UCM6108, DAHDI/(1-8) are FXO ports, and DAHDI(9-10) are FXS ports.
For UCM6116, DAHDI/(1-16) are FXO ports, and DAHDI/(17-18) are FXS ports.
Sample 2:
Figure 155: Downloaded CDR File Sample - Source Channel and Dest Channel 2
"SIP" means it's a SIP call. There are three possible format:
(a) SIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string
and can be ignored.
(c) SIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out
through the trunk.
(c) SIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this
trunk. The last XXXXX is a random string and can be ignored.
Sample 3:
Figure 156: Downloaded CDR File Sample - Source Channel and Dest Channel 3
This is a very special channel name. If it shows up, most likely it means a conference call.
There are some other possible values, but these values are almost the application name which are used
by the dialplan.
IAX2/NUM-XXXXXXX: it means this is an IAX call.
Local/@from-internal-XXXXX: it is used internally to do some special feature procedure. We can simply
ignore it.
Hangup: the call is hung up from the dialplan. This indicates there are some errors or it has run into
abnormal cases.
Playback: play some prompts to you, such as 183 response or run into an IVR.
ReadExten: collect numbers from user. It may occur when you input PIN codes or run into DISA
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STATISTICS
CDR Statistics is an additional feature on the UCM6100 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 157: CDR Statistics
Table 99: 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
UCM6100.
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Figure 158: CDR->Recording Files

Click on “Delete Selected Recording Files” to delete the recording files.

Click on “Delete All Recording Files” to delete all recording files.

Click on
to download the recording file in .wav format.

Click on
to delete the recording file.

To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding
column. Click on the title again can switch the sorting mode between ascending order or descending
order.
API CONFIGURATION
The UCM6100 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 UCM6100 first. The API configuration parameters are listed in the table below.
Table 100: API Configuration Files
Enable
Enable/Disable API. The default setting is disabled.
TLS Bind Address
Configure the IP address for TLS server to bind to. "0.0.0.0" means binding to all
interfaces. The port number is optional and the default port number is 8443. The IP
address must match the common name (host name) in the certificate so that the TLS
socket won't bind to multiple IP addresses. The default setting is 0.0.0.0:8443.
TLS Private Key
Upload TLS private key. The size of the key file must be under 2MB. This file will be
renamed as 'private.pem' automatically.
TLS Cert
Upload TLS cert. The size of the certificate must be under 2MB. This is the certificate
file (*.pem format only) for TLS connection. This file will be renamed as
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"certificate.pem" automatically. It contains private key for the client and signed
certificate for the server.
TLS Authentication
Configure the user name for TLS authentication. If not configured, authentication will
Name
be skipped.
TLS Authentication
Password
Permitted
Configure the password for TLS authentication. This is optional.
Specify a list of IP addresses permitted by API. This creates an AIP-specific access
control list. Multiple entries are allowed.
For example, "192.168.40.3/255.255.255.255" denies access from all IP addresses
except 192.168.40.3.
The default setting is blank, meaning all IPs will be denied. Users must set permitted
IP address before connecting to the API.
For more details on CDR API (Access to Call Detail Records) and REC API (Access to Call Recording
Files), please refer the document in the link here:
http://www.grandstream.com/products/ucm_series/ucm61xx/documents/ucm61xx_cdr_rec_api_guide.pdf
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UPGRADING AND MAINTENANCE
UPGRADING
The UCM6100 can be upgraded to a new firmware version remotely or locally. This section describes how
to upgrade your UCM6100 via network or local upload.
UPGRADING VIA NETWORK
The UCM6100 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
The upgrading configuration can be accessed via Web GUI->Maintenance->Upgrade.
Figure 159: Network Upgrade
Table 101: 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 UCM6100.
Firmware File Suffix
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downloaded and flashed into the UCM6100.
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 UCM6100 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 UCM6100 directly via Web
GUI. Please follow the steps below to upload firmware locally.

Download the latest UCM6100 firmware file from the following link and save it in your PC.
http://www.grandstream.com/support/firmware

Log in the Web GUI as administrator in the PC.

Go to Web GUI->Maintenance->Upgrade, upload the firmware file by clicking on
and
select the firmware file from your PC. The default firmware file name is ucm6100fw.bin
Figure 160: Local Upgrade

Click on
to start upgrading.
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Figure 161: Upgrading Firmware Files

Wait until the upgrading process is successful and a window will be popped up in the Web GUI.
Figure 162: Reboot UCM6100

Click on "OK" to reboot the UCM6100 and check the firmware version after it boots up.
Note:
Please do not interrupt or power cycle the UCM6100 during upgrading process.
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NO LOCAL FIRMWARE SERVERS
For users that would like to use remote upgrading without a local TFTP server, Grandstream offers a
NAT-friendly HTTP server. This enables users to download the latest software upgrades for their devices
via this server. Please refer to the webpage:
http://www.grandstream.com/support/firmware.
Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A
free windows version TFTP server is available for download from :
http://www.solarwinds.com/products/freetools/free_tftp_server.aspx
http://tftpd32.jounin.net
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 UCM6100 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 UCM6100 web configuration interface;
5. Configure the Firmware Server Path to the IP address of the PC;
6. Update the changes and reboot the UCM6100.
End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use
Microsoft IIS web server.
BACKUP
The UCM6100 configuration can be backed up locally or via network. The backup file will be used to
restore the configuration on UCM6100 when necessary.
LOCAL BACKUP
Users
could
backup
the
UCM6100
configurations
for
restore
purpose
under
Web
GUI->Maintenance->Backup->Local Backup. Before creating new backup file, select the backup option
first.

If the Config-File is selected only, the backup file will be saved in the flash of the UCM6100.

If Voice-File, Voicemail-File, Voice-Records, CDR or VFAX is selected, external storage devices (USB
Flash drive or SD Card) will be required because the backup file might be too large.
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Click on "Create New Backup" button 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 UCM6100 internal storage or the external device.
Figure 163: 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.
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Figure 164: Data Sync
Table 102: Data Sync Configuration
Enable Backup
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.
Backup Time
Enter 0-23 to specify the backup hour of the day.
Before saving the configuration, users could click on "Test Connection". The UCM6100 will then try
connecting the server to make sure the server is up and accessible for the UCM6100. Save the changes
and all the backup logs will be listed on the web page. After data sync is configured, users could also
manually synchronize all data by clicking on
instead of waiting for the backup time
interval to come.
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RESTORE CONFIGURATION FROM BACKUP FILE
To restore the configuration on the UCM6100 from a backup file, users could go to Web
GUI->Maintenance->Backup->Local Backup.

A list of previous configuration backups is displayed on the web page. Users could click on
of the
desired backup file and it will be restored to the UCM6100.

If users have other backup files on PC to restore on the UCM6100, click on "Upload Backup File" first
and select it from local PC to upload on the UCM6100. Once the uploading is done, this backup file will
be displayed in the list of previous configuration backups for restore purpose. Click on
to restore
from the backup file.
Figure 165: Restore UCM6100 from Backup File
Note:
 The uploaded backup file must be a tar file with no special characters like *,!,#,@,&,$,%,^,(,),/,\,space
in the file name.
 The uploaded back file size must be under 10MB.
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CLEANER
Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under
Web GUI->Maintenance->Cleaner.
Figure 166: Cleaner
Table 103: Cleaner Configuration
Enable CDR Cleaner
Enable the CDR Cleaner function.
CDR Clean Time
Enter 0-23 to specify the hour of the day to clean up CDR.
Clean Interval
Enter 1-30 to specify the day of the month to clean up CDR.
Enable VR Cleaner
Enter the Voice Records Cleaner function.
Choose Cleaner File
Select the files for system automatic clean.
VR Clean Threshold
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
Recording Files

Conference

Queue

Voicemail

Fax
Specify the Voice Records threshold from 0 to 99 by using local storage
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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.

All: All the configurations and data will be reset to factory default.
Figure 167: Reset and Reboot
SYSLOG
On the UCM6100, 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 UCM6100 settings
because it can be helpful to locate the issues when errors happen.
Some typical modules for UCM6100 functions are as follows and users can turn on "notic" and "verb"
levels besides "error" level.
pbx: This module is related to general PBX functions.
chan_sip: This module is related to SIP calls.
chan_dahdi: This module is related to analog calls (FXO/FXS).
app_meetme: This module is related to conference bridge.
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Note:
Syslog is usually for debugging and troubleshooting purpose. Turning on all levels for all syslog modules is
not recommended for daily usage. Too many syslog print might cause traffic and affect system
performance.
TROUBLESHOOTING
On the UCM6100, 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 168: 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 169: 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 170: Traceroute
ANALOG RECORD TRACE
Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM6100 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:
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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 171: Troubleshooting Analog Trunks
After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream
Technical support in the following link for further assistance if the issue is not resolved.
http://www.grandstream.com/index.php/support
SERVICE CHECK
Enable Service Check to periodically check Asterisk service. 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 Asterisk
service. The default setting is 3. If there is no response from Asterisk service after 3 attempts (default) to check,
current status will be stored and Asterisk service will be restarted.
Figure 172: Service Check
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EXPERIENCING THE UCM6100 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 UCM6100 series IP PBX appliance, it will be sure to bring
convenience and color to both your business and personal life.
* Asterisk is a Registered Trademark of Digium, Inc.
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FCC Caution:
Any Changes or modifications not expressly approved by the party responsible for compliance could void
the user's authority to operate the equipment.
This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:
(1) This device may not cause harmful interference, and (2) this device must accept any interference
received, including interference that may cause undesired operation.
Note: This equipment has been tested and found to comply with the limits for a Class B digital device,
pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against
harmful interference in a residential installation. This equipment generates, uses and can radiate radio
frequency energy and, if not installed and used in accordance with the instructions, may cause harmful
interference to radio communications. However, there is no guarantee that interference will not occur in a
particular installation. If this equipment does cause harmful interference to radio or television reception,
which can be determined by turning the equipment off and on, the user is encouraged to try to correct the
interference by one or more of the following measures:
- Reorient or relocate the receiving antenna.
- Increase the separation between the equipment and receiver.
- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.
- Consult the dealer or an experienced radio/TV technician for help.
Regulatory Information
U.S. FCC Part 68 Statement
This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. The
unit bears a label on the back which contains among other information a product identifier in the format US:
GNIIS00BUCM6104/US: GNIIS00BUCM6116. If requested, this number must be provided to the
telephone company.
This equipment uses the following standard jack types for network connection: RJ11C.
This equipment contains an FCC compliant modular jack. It is designed to be connected to the telephone
network or premises wiring using compatible modular plugs and cabling which comply with the
requirements of FCC Part 68 rules.
The Ringer Equivalence Number, or REN, is used to determine the number of devices which may be
connected to the telephone line. An excessive REN may cause the equipment to not ring in response to an
incoming call. In most areas, the sum of the RENs of all equipment on a line should not exceed five (5.0).
In the unlikely event that this equipment causes harm to the telephone network, the telephone company
can temporarily disconnect your service. The telephone company will try to warn you in advance of any
such disconnection, but if advance notice isn't practical, it may disconnect the service first and notify you
as soon as possible afterwards. In the event such a disconnection is deemed necessary, you will be
advised of your right to file a complaint with the FCC.
From time to time, the telephone company may make changes in its facilities, equipment, or operations
which could affect the operation of this equipment. If this occurs, the telephone company is required to
provide you with advance notice so you can make the modifications necessary to obtain uninterrupted
service.
There are no user serviceable components within this equipment. See Warranty flyer for repair or warranty
information.
It shall be unlawful for any person within the United States to use a computer or other electronic device to
send any message via a telephone facsimile unless such message clearly contains, in a margin at the top
or bottom of each transmitted page or on the first page of the transmission, the date and time it is sent and
an identification of the business, other entity, or individual sending the message and the telephone number
of the sending machine or of such business, other entity, or individual. The telephone number provided
may not be a 900 number or any other number for which charges exceed local or long distance
transmission charges. Telephone facsimile machines manufactured on and after December 20, 1992, must
clearly mark such identifying information on each transmitted message. Facsimile modem boards
manufactured on and after December 13, 1995, must comply with the requirements of this section.
This equipment cannot be used on public coin phone service provided by the telephone company.
Connection to Party Line Service is subject to state tariffs. Contact your state public utility commission,
public service commission, or corporation commission for more information.
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