Grandstream UCM6116 User Manual

Grandstream UCM6116 User Manual
Grandstream Networks, Inc.
UCM6100 Series IP PBX
User Manual
UCM6100 SERIES IP PBX USER MANUAL
UCM6100 Series IP PBX User Manual
Index
CHANGE LOG ........................................................................................... 12
FIRMWARE VERSION 1.0.6.10 .......................................................................................................... 12
FIRMWARE VERSION 1.0.5.19 .......................................................................................................... 12
FIRMWARE VERSION 1.0.5.14 .......................................................................................................... 13
FIRMWARE VERSION 1.0.4.7 ............................................................................................................ 13
FIRMWARE VERSION 1.0.3.13 .......................................................................................................... 14
FIRMWARE VERSION 1.0.2.21 .......................................................................................................... 14
FIRMWARE VERSION 1.0.1.22 .......................................................................................................... 15
WELCOME ................................................................................................ 16
PRODUCT OVERVIEW ............................................................................. 17
FEATURE HIGHTLIGHTS ................................................................................................................... 17
TECHNICAL SPECIFICATIONS .......................................................................................................... 17
INSTALLATION ......................................................................................... 20
EQUIPMENT PACKAGING ................................................................................................................. 20
CONNECT YOUR UCM6100............................................................................................................... 20
CONNECT THE UCM6102........................................................................................................... 20
CONNECT THE UCM6104........................................................................................................... 22
CONNECT THE UCM6108........................................................................................................... 22
CONNECT THE UCM6116 ........................................................................................................... 23
SAFETY COMPLIANCES .................................................................................................................... 24
WARRANTY......................................................................................................................................... 24
GETTING STARTED ................................................................................. 24
USE THE LCD MENU.......................................................................................................................... 25
USE THE LED INDICATORS .............................................................................................................. 27
USE THE WEB GUI ............................................................................................................................. 28
ACCESS WEB GUI ...................................................................................................................... 28
WEB GUI CONFIGURATIONS ..................................................................................................... 29
WEB GUI LANGUAGES ............................................................................................................... 29
SAVE AND APPLY CHANGES ..................................................................................................... 30
MAKE YOUR FIRST CALL .................................................................................................................. 31
SYSTEM SETTINGS ................................................................................. 32
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NETWORK SETTINGS........................................................................................................................ 32
BASIC SETTINGS ........................................................................................................................ 32
802.1X .......................................................................................................................................... 36
PORT FORWORDING (UCM6102 ONLY) ................................................................................... 37
STATIC ROUTES ......................................................................................................................... 37
FIREWALL ........................................................................................................................................... 38
STATIC DEFENSE ....................................................................................................................... 38
DYNAMIC DEFENSE ................................................................................................................... 41
FAIL2BAN ..................................................................................................................................... 41
CHANGE PASSWORD ........................................................................................................................ 42
LDAP SERVER .................................................................................................................................... 43
LDAP SERVER CONFIGURATIONS ........................................................................................... 43
LDAP PHONEBOOK .................................................................................................................... 44
LDAP CLIENT CONFIGURATIONS ............................................................................................. 46
HTTP SERVER .................................................................................................................................... 47
EMAIL SETTINGS ............................................................................................................................... 48
TIME SETTINGS ................................................................................................................................. 49
NTP SERVER ...................................................................................................................................... 51
PROVISIONING ......................................................................................... 52
OVERVIEW .......................................................................................................................................... 52
AUTO PROVISIONING........................................................................................................................ 52
MANUAL PROVISIONING ................................................................................................................... 55
DISCOVERY ................................................................................................................................. 55
ASSIGNMENT .............................................................................................................................. 56
CREATE NEW DEVICE................................................................................................................ 57
PROVISIONING ........................................................................................................................... 57
EXTENSIONS ............................................................................................ 58
CREATE NEW USER .......................................................................................................................... 58
CREATE NEW SIP EXTENSION ................................................................................................. 58
CREATE NEW IAX EXTENSION ................................................................................................. 61
CREATE NEW FXS EXTENSION ................................................................................................ 64
BATCH ADD EXTENSIONS ................................................................................................................ 67
BATCH ADD SIP EXTENSIONS .................................................................................................. 68
BATCH ADD IAX EXTENSIONS .................................................................................................. 70
EDIT EXTENSION ............................................................................................................................... 73
EXPORT EXTENSIONS ...................................................................................................................... 74
IMPORT EXTENSIONS ....................................................................................................................... 74
TRUNKS .................................................................................................... 76
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ANALOG TRUNKS .............................................................................................................................. 76
ANALOG TRUNK CONFIGURATION .......................................................................................... 76
PSTN DETECTION ...................................................................................................................... 78
VOIP TRUNKS ..................................................................................................................................... 81
Direct Outward Dialing (DOD) ...................................................................................................... 89
CALL ROUTES.......................................................................................... 92
OUTBOUND ROUTES ........................................................................................................................ 92
INBOUND ROUTES ............................................................................................................................ 94
INBOUND RULE CONFIGURATIONS ......................................................................................... 95
BLACKLIST CONFIGURATIONS ................................................................................................. 97
CONFERENCE BRIDGE ........................................................................... 99
CONFERENCE BRIDGE CONFIGURATIONS ............................................................................ 99
JOIN A CONFERENCE CALL .................................................................................................... 101
INVITE OTHER PARTIES TO JOIN CONFERENCE ................................................................. 101
DURING THE CONFERENCE ................................................................................................... 102
RECORD CONFERENCE .......................................................................................................... 103
IVR ........................................................................................................... 105
CONFIGURE IVR .............................................................................................................................. 105
CREATE IVR PROMPT ..................................................................................................................... 107
RECORD NEW IVR PROMPT ................................................................................................... 107
UPLOAD IVR PROMPT ............................................................................................................. 108
LANGUAGE SETTINGS FOR VOICE PROMPT .................................... 109
DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE ............................................................... 109
CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE................................................................111
VOICEMAIL ............................................................................................. 112
CONFIGURE VOICEMAIL ................................................................................................................. 112
VOICEMAIL EMAIL SETTINGS ......................................................................................................... 113
CONFIGURE VOICEMAIL GROUP................................................................................................... 114
RING GROUP .......................................................................................... 116
CONFIGURE RING GROUP ............................................................................................................. 116
PAGING AND INTERCOM GROUP ........................................................ 118
CONFIGURE PAGING/INTERCOM GROUP .................................................................................... 118
CALL QUEUE .......................................................................................... 120
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CONFIGURE CALL QUEUE.............................................................................................................. 120
EXTENSION GROUPS ............................................................................ 124
CONFIGURE EXTENSION GROUPS ............................................................................................... 124
USE EXTENSION GROUPS ............................................................................................................. 125
PICKUP GROUPS ................................................................................... 126
CONFIGURE PICKUP GROUPS ...................................................................................................... 126
MUSIC ON HOLD .................................................................................... 127
FAX/T.38 .................................................................................................. 128
CONFIGURE FAX/T.38 ...................................................................................................................... 128
SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE ............................................... 129
SAMPLE CONFIGURATION FOR FAX-TO-EMAIL........................................................................... 131
DISA ........................................................................................................ 133
BLF AND EVENT LIST ............................................................................ 135
BLF..................................................................................................................................................... 135
EVENT LIST....................................................................................................................................... 135
DIAL BY NAME ....................................................................................... 138
DIAL BY NAME CONFIGURATION ................................................................................................... 138
CALL FEATURES ................................................................................... 142
FEATURE CODES ............................................................................................................................. 142
CALL RECORDING ........................................................................................................................... 145
CALL PARK ........................................................................................................................................ 146
PARK A CALL ............................................................................................................................. 146
RETRIEVE THE PARKED CALL ................................................................................................ 146
INTERNAL OPTIONS .............................................................................. 147
INTERNAL OPTIONS/GENERAL ...................................................................................................... 147
INTERNAL OPTIONS/JITTER BUFFER ........................................................................................... 148
INTERNAL OPTIONS/RTP SETTINGS ............................................................................................. 149
INTERNAL OPTIONS/PORTS CONFIG............................................................................................ 149
INTERNAL OPTIONS/STUN MONITOR ........................................................................................... 151
INTERNAL OPTIONS/PAYLOAD....................................................................................................... 152
IAX SETTINGS ........................................................................................ 153
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IAX SETTINGS/GENERAL ................................................................................................................ 153
IAX SETTINGS/REGISTRATION ...................................................................................................... 153
IAX SETTINGS/STATIC DEFENSE ................................................................................................... 154
SIP SETTINGS ........................................................................................ 156
SIP SETTINGS/GENERAL ................................................................................................................ 156
SIP SETTINGS/MISC ........................................................................................................................ 157
SIP SETTINGS/SESSION TIMER ..................................................................................................... 157
SIP SETTINGS/TCP and TLS ........................................................................................................... 158
SIP SETTINGS/NAT .......................................................................................................................... 159
SIP SETTINGS/TOS .......................................................................................................................... 160
STATUS AND REPORTING .................................................................... 162
PBX STATUS ..................................................................................................................................... 162
TRUNKS ..................................................................................................................................... 162
EXTENSIONS............................................................................................................................. 163
QUEUES..................................................................................................................................... 165
CONFERENCE ROOMS ............................................................................................................ 166
INTERFACES STATUS .............................................................................................................. 166
PARKING LOT ............................................................................................................................ 167
ACTIVITY CALLS .............................................................................................................................. 168
SYSTEM STATUS.............................................................................................................................. 169
GENERAL................................................................................................................................... 169
NETWORK ................................................................................................................................. 170
STORAGE USAGE..................................................................................................................... 170
RESOURCE USAGE .................................................................................................................. 171
SYSTEM EVENTS ............................................................................................................................. 172
ALERT EVENTS LIST ................................................................................................................ 172
ALERT LOG ................................................................................................................................ 174
ALERT CONTACT ...................................................................................................................... 174
CDR ................................................................................................................................................... 174
DOWNLOADED CDR FILE ........................................................................................................ 176
STATISTICS................................................................................................................................ 178
RECORDING FILES ................................................................................................................... 179
CDR API CONFIGURATION FILES ........................................................................................... 180
UPGRADING AND MAINTENANCE ....................................................... 187
UPGRADING ..................................................................................................................................... 187
UPGRADING VIA NETWORK .................................................................................................... 187
UPGRADING VIA LOCAL UPLOAD........................................................................................... 188
NO LOCAL FIRMWARE SERVERS ........................................................................................... 190
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BACKUP ............................................................................................................................................ 190
LOCAL BACKUP ........................................................................................................................ 190
DATA SYNC ................................................................................................................................ 191
RESTORE CONFIGURATION FROM BACKUP FILE ............................................................... 192
CLEANER .......................................................................................................................................... 193
RESET AND REBOOT ...................................................................................................................... 194
SYSLOG ............................................................................................................................................ 195
TROUBLESHOOTING ....................................................................................................................... 195
ETHERNET CAPTURE .............................................................................................................. 196
IP PING ....................................................................................................................................... 196
TRACEROUTE ........................................................................................................................... 197
EXPERIENCING THE UCM6100 SERIES IP PBX .................................. 198
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Table of Tables
UCM6100 Series IP PBX User Manual
Table 1: Technical Specifications ................................................................................................................ 17
Table 2: UCM6102/UCM6104 Equipment Packaging ................................................................................. 20
Table 3: UCM6108/UCM6116 Equipment Packaging ................................................................................. 20
Table 4: LCD Menu Options ........................................................................................................................ 26
Table 5: UCM6102/UCM6104 LED INDICATORS ...................................................................................... 27
Table 6: UCM6108/UCM6116 LED INDICATORS ...................................................................................... 27
Table 7: UCM6102 Network Settings->Basic Settings ................................................................................ 32
Table 8: UCM6104 Network Settings->Basic Settings ................................................................................ 34
Table 9: UCM6108/UCM6116 Network Settings->Basic Settings ............................................................... 36
Table 10: UCM6100 Network Settings->802.1X ......................................................................................... 36
Table 11: UCM6102 Network Settings->Port Forwarding ........................................................................... 37
Table 12: UCM6100 Network Settings->Static Routes ............................................................................... 37
Table 13: UCM6100 Firewall->Static Defense->Current Service ................................................................ 39
Table 14: Typical Firewall Settings .............................................................................................................. 39
Table 15: Firewall Rule Settings .................................................................................................................. 40
Table 16: UCM6102 Firewall Dynamic Defense ......................................................................................... 41
Table 17: Fail2Ban Settings ........................................................................................................................ 42
Table 18: HTTP Server Settings.................................................................................................................. 48
Table 19: Email Settings.............................................................................................................................. 48
Table 20: Time Auto Updating ..................................................................................................................... 50
Table 21: Auto Provision Settings ............................................................................................................... 54
Table 22: SIP Extension Configuration Parameters .................................................................................... 58
Table 23: IAX Extension Configuration Parameters .................................................................................... 62
Table 24: FXS Extension Configuration Parameters .................................................................................. 64
Table 25: Batch Add SIP Extension Parameters ......................................................................................... 68
Table 26: Batch Add IAX Extension Parameters ......................................................................................... 71
Table 27: Analog Trunk Configuration Parameters ..................................................................................... 76
Table 28: PSTN Detection For Analog Trunk .............................................................................................. 81
Table 29: SIP Trunk Configuration Parameters........................................................................................... 82
Table 30: IAX Trunk Configuration Parameters........................................................................................... 87
Table 31: Outbound Route Configuration Parameters ................................................................................ 92
Table 32: Inbound Rule Configuration Parameters ..................................................................................... 95
Table 33: Conference Bridge Configuration Parameters ............................................................................ 99
Table 34: Conference Caller IVR Menu .................................................................................................... 103
Table 35: IVR Configuration Parameters .................................................................................................. 105
Table 36: Voicemail Settings ..................................................................................................................... 112
Table 37: Voicemail Email Settings ........................................................................................................... 114
Table 38: Voicemail Group Settings .......................................................................................................... 115
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Table 39: Ring Group Parameters ............................................................................................................ 116
Table 40: Paging/Intercom Group Configuration Parameters ................................................................... 118
Table 41: Call Queue Configuration Parameters ...................................................................................... 120
Table 42: FAX/T.38 Settings ...................................................................................................................... 128
Table 43: DISA Settings ............................................................................................................................ 133
Table 44: Event List Settings ..................................................................................................................... 136
Table 45: UCM6100 Feature Codes ......................................................................................................... 142
Table 46: Internal Options/General ........................................................................................................... 147
Table 47: Internal Options/Jitter Buffer ...................................................................................................... 149
Table 48: Internal Options/RTP Settings ................................................................................................... 149
Table 49: Internal Options/Ports Config .................................................................................................... 150
Table 50: Internal Options/STUN Monitor ................................................................................................. 151
Table 51: Internal Options/Payload ........................................................................................................... 152
Table 52: IAX Settings/General ................................................................................................................. 153
Table 53: IAX Settings/Registration .......................................................................................................... 153
Table 54: IAX Settings/Static Defense ...................................................................................................... 154
Table 55: SIP Settings/General ................................................................................................................. 156
Table 56: SIP Settings/Misc ...................................................................................................................... 157
Table 57: SIP Settings/Session Timer ....................................................................................................... 157
Table 58: SIP Settings/TCP and TLS ........................................................................................................ 158
Table 59: SIP Settings/NAT ....................................................................................................................... 159
Table 60: SIP Settings/ToS........................................................................................................................ 160
Table 61: Trunk Status............................................................................................................................... 163
Table 62: Extension Status ........................................................................................................................ 164
Table 63: Agent Status .............................................................................................................................. 165
Table 64: Interface Status Indicators ......................................................................................................... 167
Table 65: Parking Lot Status ..................................................................................................................... 168
Table 66: System Status->General ........................................................................................................... 169
Table 67: System Status->Network ........................................................................................................... 170
Table 68: CDR Filter Criteria ..................................................................................................................... 175
Table 69: CDR Statistics Filter Criteria ...................................................................................................... 179
Table 70: CDR API Configuration Files ..................................................................................................... 180
Table 71: CDR API URI Parameters ......................................................................................................... 181
Table 72: Network Upgrade Configuration ................................................................................................ 187
Table 73: Data Sync Configuration ........................................................................................................... 192
Table 74: Cleaner Configuration ............................................................................................................... 194
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Table of Figures
UCM6100 Series IP PBX User Manual
Figure 1: UCM6102 Front View................................................................................................................... 21
Figure 2: UCM6102 Back View ................................................................................................................... 21
Figure 3: UCM6104 Front View................................................................................................................... 22
Figure 4: UCM6104 Back View ................................................................................................................... 22
Figure 5: UCM6108 Front View................................................................................................................... 23
Figure 6: UCM6108 Back View ................................................................................................................... 23
Figure 7: UCM6116 Front View ................................................................................................................... 23
Figure 8: UCM6116 Back View ................................................................................................................... 23
Figure 9: UCM6116 Web GUI Login Page .................................................................................................. 28
Figure 10: UCM6100 Web GUI Language .................................................................................................. 30
Figure 11: Create New Firewall Rule .......................................................................................................... 40
Figure 12: LDAP Server Configurations ...................................................................................................... 43
Figure 13: Default LDAP Phonebook DN .................................................................................................... 44
Figure 14: Default LDAP Phonebook Attributes .......................................................................................... 44
Figure 15: Add LDAP Phonebook ............................................................................................................... 45
Figure 16: Edit LDAP Phonebook ............................................................................................................... 45
Figure 17: GXP2200 LDAP Phonebook Configuration ............................................................................... 47
Figure 18: UCM6100 Email Settings ........................................................................................................... 49
Figure 19: Set Time Manually ..................................................................................................................... 51
Figure 20: UCM6100 Zero Config ............................................................................................................... 53
Figure 21: Auto Provision Settings .............................................................................................................. 54
Figure 22: Auto Discover ............................................................................................................................. 55
Figure 23: Discovered Devices ................................................................................................................... 56
Figure 24: Assign Extension To Device ....................................................................................................... 56
Figure 25: Create New Device .................................................................................................................... 57
Figure 26: Export Extensions ...................................................................................................................... 74
Figure 27: Import Extensions ...................................................................................................................... 74
Figure 28: UCM6100 FXO Tone Settings ................................................................................................... 79
Figure 29: UCM6100 PSTN Detection ........................................................................................................ 79
Figure 30: UCM6100 PSTN Detection: Auto Detect ................................................................................... 80
Figure 31: UCM6100 PSTN Detection: Semi-Auto Detect ......................................................................... 80
Figure 32: DOD extension selection ........................................................................................................... 90
Figure 33: Edit DOD .................................................................................................................................... 91
Figure 34: Blacklist Configuration Parameters............................................................................................ 97
Figure 35: Conference Invitation From Web GUI...................................................................................... 101
Figure 36: Conference Recording ............................................................................................................. 104
Figure 37: Click On Prompt To Create IVR Prompt .................................................................................. 107
Figure 38: Record New IVR Prompt ......................................................................................................... 107
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Figure 39: Upload IVR Prompt .................................................................................................................. 108
Figure 40: Language Settings For Voice Prompt ...................................................................................... 110
Figure 41: Voice Prompt Package List ...................................................................................................... 110
Figure 42: New Voice Prompt Language Added ........................................................................................111
Figure 43: Voicemail Email Settings ......................................................................................................... 113
Figure 44: Voicemail Group....................................................................................................................... 114
Figure 45: Ring Group............................................................................................................................... 116
Figure 46: Ring Group Configuration ........................................................................................................ 117
Figure 47: Paging/Intercom Group............................................................................................................ 118
Figure 48: Page/Intercom Group Settings ................................................................................................ 119
Figure 49: Call Queue ............................................................................................................................... 120
Figure 50: Agent Login Settings ................................................................................................................ 122
Figure 51: Edit Extension Group ............................................................................................................... 124
Figure 52: Select Extension Group in Outbound Route............................................................................ 125
Figure 53: Edit Pickup Group .................................................................................................................... 126
Figure 54: Music On Hold Default Class ................................................................................................... 127
Figure 55: Configure Analog Trunk without Fax Detection ....................................................................... 129
Figure 56: Configure Extension For Fax Machine .................................................................................... 130
Figure 57: Configure Inbound Rule For Fax ............................................................................................. 130
Figure 58: Create Fax Extension .............................................................................................................. 131
Figure 59: Inbound Route To Fax Extension............................................................................................. 131
Figure 60: Create New DISA..................................................................................................................... 133
Figure 61: Create New Event List ............................................................................................................. 136
Figure 62: Create Dial By Name Group .................................................................................................... 138
Figure 63: Dial By Name Group In IVR Key Pressing Events .................................................................. 139
Figure 64: Dial By Name Group In Inbound Rule ..................................................................................... 140
Figure 65: Configure Extension First Name And Last Name .................................................................... 141
Figure 66: Download Recording File From CDR Page ............................................................................. 145
Figure 67: FXS Ports Signaling Preference .............................................................................................. 150
Figure 68: FXO Ports ACIM Settings ........................................................................................................ 150
Figure 69: Status->PBX Status ................................................................................................................. 162
Figure 70: Trunk Status ............................................................................................................................. 162
Figure 71: Extension Status ...................................................................................................................... 164
Figure 72: Queue Status ........................................................................................................................... 165
Figure 73: Conference Room Status ......................................................................................................... 166
Figure 74: UCM6116 Interfaces Status ..................................................................................................... 166
Figure 75: Parking Lot Status .................................................................................................................... 167
Figure 76: Status->PBX Status->Activity Calls: Calling ............................................................................ 168
Figure 77: Status->PBX Status->Activity Calls.......................................................................................... 169
Figure 78: System Status->Storage Usage ............................................................................................... 171
Figure 79: System Status->Resource Usage ............................................................................................ 171
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Figure 80: System Events->Alert Events Lists: Disk Usage ..................................................................... 172
Figure 81: System Events->Alert Events Lists: Memory Usage ............................................................... 173
Figure 82: System Events->Alert Events Lists: System Reboot ............................................................... 173
Figure 83: System Events->Alert Events Lists: System Crash ................................................................. 173
Figure 84: System Events->Alert Log ....................................................................................................... 174
Figure 85: CDR Filter ................................................................................................................................ 175
Figure 86: Call Report ............................................................................................................................... 176
Figure 87: Call Report Entry With Audio Recording File ........................................................................... 176
Figure 88: Downloaded CDR File Sample - Call To Shows "s" ................................................................ 177
Figure 89: Downloaded CDR File Sample - Source Channel and Dest Channel 1 .................................. 177
Figure 90: Downloaded CDR File Sample - Source Channel and Dest Channel 2 .................................. 178
Figure 91: Downloaded CDR File Sample - Source Channel and Dest Channel 3 .................................. 178
Figure 92: CDR Statistics .......................................................................................................................... 179
Figure 93: CDR->Recording Files ............................................................................................................. 180
Figure 94: Network Upgrade ..................................................................................................................... 187
Figure 95: Local Upgrade.......................................................................................................................... 188
Figure 96: Upgrading Firmware Files ........................................................................................................ 189
Figure 97: Reboot UCM6100 .................................................................................................................... 189
Figure 98: Local Backup ........................................................................................................................... 191
Figure 99: Data Sync ................................................................................................................................ 192
Figure 100: Restore UCM6100 From Backup File .................................................................................... 193
Figure 101: Cleaner .................................................................................................................................. 194
Figure 102: Reset and Reboot .................................................................................................................. 195
Figure 103: Ethernet Capture ................................................................................................................... 196
Figure 104: PING ...................................................................................................................................... 196
Figure 105: Traceroute .............................................................................................................................. 197
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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 1.0.6.10

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.

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. [ACTIVITY CALLS]

Added CDR API support. [CDR API CONFIGURATION FILES]

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 45: UCM6100 Feature Codes]

Changed valid range for option "Current Disconnect Threshold". [Table 27: Analog Trunk
Configuration Parameters]
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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 31: 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 39: 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 46: Internal Options/General]

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 29: SIP Trunk Configuration Parameters]

Added DOD routing support. [Direct Outward Dialing (DOD)]

Added GXP one-button Voicemail access. [Table 22: SIP Extension Configuration Parameters]

Added option "Skip voicemail password verification" on extension edit page. [Table 22: SIP
Extension Configuration Parameters]

Added Hot-Desking Support. [Table 22: 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 49:
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 "Pick Extension Period" on auto-provision settings page of Zero Config. [Table 21: Auto
Provision Settings]

Added multiple extension assignment support on device edit page of Zero Config. [Figure 24:
Assign Extension To Device]

Added "Reset All Extensions" button at the Zero Config page to recycle all assigned extensions.
[AUTO PROVISIONING]
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
Added system crash alarm, core dump detection and allow users to download core dump file.
[Figure 83: 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 29: SIP Trunk Configuration Parameters]
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 22: SIP
Extension Configuration Parameters] [Table 25: Batch Add SIP Extension Parameters]

Added "Auto Record" option for trunk. [Table 27: Analog Trunk Configuration Parameters] [Table 29:
SIP Trunk Configuration Parameters]
Added support to specify caller ID in outbound route and inbound route. [Table 31: Outbound Route Configuration
Parameters] [

Table 32: Inbound Rule Configuration Parameters]

Added "Digit Timeout" option and voice prompt "Language" selection for IVR. [Table 35: IVR
Configuration Parameters]

Added "Direct Dial Voicemail Prefix" feature code to directly dial or transfer to extension's voicemail.
[Table 45: UCM6100 Feature Codes]

Added "Enforce Strong Passwords" option. [Table 46: Internal Options/General]

Added FXS MWI Mode. [Table 49: Internal Options/Ports Config]

Added system events with alert and Email notification support. [SYSTEM EVENTS]

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. [LDAP CLIENT CONFIGURATIONS]

Added sample Email settings. [EMAIL SETTINGS]

Added manual time settings. [TIME SETTINGS]

Added "Enable Pick Extension" and "Extension Segment" options for auto provisioning settings. [Table
21: Auto Provision Settings]

Changed one of the discovery method from "SIP MESSAGE (OPTIONS)" to "SIP MESSAGE
(NOTIFY)" in zero-config feature. [DISCOVERY]
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
Added pickup group feature. [Table 45: 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 29: SIP Trunk Configuration
Parameters]

Added LDAP sync options for peer SIP trunk. [Table 29: SIP Trunk Configuration Parameters]

Changed the default setting of outbound route "Privilege Level" from "Internal" to "International" to
avoid potential misconfiguration and security risk. [Table 31: Outbound Route Configuration
Parameters]
Added DISA and Fax to inbound route default destination options. [

Table 32: Inbound Rule Configuration Parameters]

Added DISA and Fax to IVR key press event options. [Table 35: IVR Configuration Parameters]

Added "Min Message Time" option in voicemail settings. [Table 36: Voicemail Settings]

Added Fax setting samples. [FAX/T.38]

Added DISA support for inbound route and IVR. [DISA]

Added Event List support to monitor local extensions and remote extensions. [BLF AND EVENT LIST]

Added feature code *0 for "Disconnect". [Table 45: UCM6100 Feature Codes]

Added feature code *8 for "Pickup Extension" in pickup group feature. [Table 45: UCM6100 Feature
Codes]

Added "Record Prompt" and "Custom Name of Pickup Group" options in internal options. [Table 46:
Internal Options/General]

Added warning information for "Allow Guest Call" option to avoid potential security risk caused by
misconfiguration. [Table 52: IAX Settings/General]

Changed reset mode to two mode "User Data" and "All". [RESET AND REBOOT]
FIRMWARE VERSION 1.0.1.22

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.
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
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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
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
Security
Media
SRTP, TLS, HTTPS, SSH
Physical
Universal Power Supply
Environmental
Dimensions
Weight
Mounting
Firmware Version 1.0.6.10

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
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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
park, call
forward,
call
transfer,
DND,
ring/hunt
group,
paging/intercom and etc

FCC: Part 15 (CFR 47) Class B, Part 68

CE: EN55022 Class B, EN55024, EN61000-3-2, EN61000-3-3,
EN60950-1, TBR21, RoHS

A-TICK: AS/NZS CISPR 22 Class B, AS/NZS CISPR 24, AS/NZS
60950, AS/ACIF S002 adITU-T K.21 (Basic Level)

Firmware Version 1.0.6.10
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)
Table 3: UCM6108/UCM6116 Equipment Packaging
Main Case
Yes (1)
Power Adaptor
Yes (1)
Ethernet Cable
Yes (1)
Quick Installation Guide
Yes (1)
Wall Mount
Yes (2)
Screws
Yes (6)
CONNECT YOUR UCM6100
CONNECT THE UCM6102
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Figure 1: UCM6102 Front View
Figure 2: 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 3: UCM6104 Front View
Figure 4: 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.
CONNECT THE UCM6108
To set up the UCM6108, follow the steps below:
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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 5: UCM6108 Front View
Figure 6: UCM6108 Back View
CONNECT THE UCM6116
Figure 7: UCM6116 Front View
Figure 8: UCM6116 Back View
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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.
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.
The following table shows the LCD menu options.
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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:
Network Info

LAN Mode: DHCP, Static IP, or PPPoE

LAN IP: IP address

LAN Subnet Mask
For UCM6102:

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:

LAN Mode: Select LAN mode as DHCP, Static IP or PPPoE
Network Menu
For UCM6102:

WAN Mode: Select WAN mode as DHCP, Static IP or PPPoE

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 9: 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
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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.
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 network settings, firewall settings, change password, LDAP Server, HTTP
Server, Email Settings, Time Settings and NTP server.

Maintenance: To perform firmware upgrade, backup configurations, cleaner setup, reset/reboot,
syslog setup and troubleshooting.
WEB GUI LANGUAGES
Currently the UCM6100 series web GUI supports the following languages:
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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.
Figure 10: 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.
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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. Those parameters
include Network Settings, Firewall, Change Password, LDAP server, HTTP server, Email settings, Time
Settings and NTP Server settings.
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 7: 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
Method
be used to serve as router.

Switch
WAN port interface will be used for uplink connection. LAN port interface will
be used as bridge for PC connection.

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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.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
Subnet Mask
IP Address
DNS Server 1
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
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.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
Subnet Mask
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
IP Address
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
DNS Server 1
Enter the DNS server 1 address for static IP settings. The default setting is
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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
802.1Q/VLAN Tag
is 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")
If "Dual" is selected as "Method", users will need assign the default interface to
Default Interface
be 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
Gateway IP
IP Address
Subnet Mask
DNS Server 1
Select DHCP, Static IP, or PPPoE. The default setting is DHCP.
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 IP address for static IP settings. The default setting is 192.168.0.160.
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
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
802.1Q/VLAN Tag
is 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 8: UCM6104 Network Settings->Basic Settings
Select "Switch" or "Dual" mode on the network interface of UCM6104. The
default setting is "Switch".
Method

Switch
LAN 1 port interface will be used for uplink connection. LAN 2 port interface
will be used as bridge for PC connection.

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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.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
Subnet Mask
IP Address
DNS Server 1
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
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
802.1Q/VLAN Tag
is 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.
Gateway IP
Enter the gateway IP address for static IP settings. The default setting is 0.0.0.0.
Subnet Mask
IP Address
DNS Server 1
Enter the subnet mask address for static IP settings. The default setting is
255.255.0.0.
Enter the IP address for static IP settings. The default setting is 192.168.0.160.
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
802.1Q/VLAN Tag
is 0.
Layer 2 QoS 802.1p
Assign the priority value of the layer 2 QoS packets for LAN port. The default
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Priority Value
value is 0.
Table 9: 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.
Gateway IP
Enter the gateway IP address for static IP settings.
Subnet Mask
Enter the subnet mask address for static IP settings.
IP Address
Enter the 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.
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
802.1Q/VLAN Tag
is 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.
802.1X
The UCM6100 provides users 802.1X settings for LAN port and WAN port (WAN port: UCM6102 only).
Table 10: 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
Certificate
Firmware Version 1.0.6.10
Select 802.1X client certificate from local PC and then upload.
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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 11: 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.
Select protocol type "UDP Only", "TCP Only" or "TCP/UDP" for the forwarding in
Protocol Type
the selected port. The default setting is "UDP Only".
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 12: UCM6100 Network Settings->Static Routes
Configure the destination IP address or the destination IP subnet for the
Destination
UCM6100 to reach using the static route.
Example:
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IP address - 192.168.66.4
IP subnet - 192.168.66.0
Configure the subnet mask for the above destination address. If left blank, the
default value is 255.255.255.255.
Netmask
Example:
255.255.255.0
Configure the gateway address so that the 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.
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.
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Table 13: UCM6100 Firewall->Static Defense->Current Service
Port
Process
Type
Protocol or Service
7777
Asterisk
tcp/IPv4
SIP
389
Slapd
tcp/IPv4
LDAP
22
Dropbear
tcp/IPv4
SSH
80
Lighthttpd
tcp/IPv4
HTTP
8089
Lighthttpd
tcp/IPv4
HTTPS
69
Opentftpd
udp/IPv4
TFTP
9090
Asterisk
udp/IPv4
SIP
6060
zero_config
udp/IPv4
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 14: 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.
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Figure 11: Create New Firewall Rule
Table 15: 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 selected, users will need specify Source (IP and port), Destination (IP
and port) and Protocol (TCP, UDP or Both) for the service.
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. Users can click on
to edit the rule, or select
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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 16: 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
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.
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Table 17: 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
MaxRetry
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.
Configure the number of authentication failures during "Max Retry Duration"
before the host is banned. The default setting is 10.
Configure IP address, CIDR mask or DNS host in the whiltelist. Fail2Ban will not
Fail2Ban Whitelist
ban the host with matching address in this list. Up to 5 addresses can be added
into the list.
Local Settings
Enable Asterisk service for Fail2Ban. The default setting is disabled. Please
Asterisk Service
make sure both "Enable Fail2Ban" and "Asterisk Service" are turned on in order
to use Fail2Ban for SIP authentication on the UCM6100.
Protocol
Configure the listening port number for the service. Currently only 5060 (for
UDP) is supported.
Configure the number of authentication failures during "Max Retry Duration"
MaxRetry
before the host is banned. The default setting is 10. Please make sure this option
is properly configured as it will override the "MaxRetry" value under "Global
Settings".
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.
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.
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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.

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 12: LDAP Server Configurations
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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.
Figure 13: Default LDAP Phonebook DN
Figure 14: Default LDAP Phonebook Attributes
LDAP PHONEBOOK
Users could use the default phonebook, edit the default phonebook as well as add new phonebook on the
LDAP server. The first phonebook with default phonebook dn "ou=pbx,dc=pbx,dc=com" displayed on the
LDAP server page is for extensions in this PBX. Users cannot add or delete contacts directly. The contacts
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information will need to be modified via Web GUI->PBX->Basic/Call Routes->Extensions first. The
default LDAP phonebook will then be updated automatically.
A new sibling phonebook of the default PBX phonebook can be added by clicking on "Add" under "LDAP
Phonebook" section.
Figure 15: 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.
Figure 16: Edit LDAP Phonebook
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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 12: LDAP Server Configurations.
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Figure 17: 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.
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Table 18: HTTP Server Settings
Enable or disable redirect from port 80. On the PBX, the default access
Redirect From Port 80
protocol is HTTPS and the default port number is 8089. When this option
is enabled, the access using HTTP with Port 80 will be redirected to
HTTPS with Port 8089. The default setting is "Enable".
Protocol Type
Port
Select HTTP or HTTPS. The default setting is "HTTPS".
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 19: 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 server for it or no user login is required. However, the
Type
Emails sent from MTA might be considered as spam by the target
SMTP server.

Client: Submit Emails to the SMTP server. A SMTP server is
required and users need login with correct credentials.
Domain
Server
Username
Password
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Specify the domain name to be used in the Email when using type
"MTA".
Specify the SMTP server when using type "Client".
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".
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Display Name
Sender
Specify the display name in the FROM header in the Email.
Specify the sender's Email address.
For example, [email protected]
The following figure shows a sample Email settings on the UCM6100, assuming the Email is using
smtp.gmail.com as the SMTP server.
Figure 18: 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
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.
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Table 20: 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
Enable DHCP Option 42
Server 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
Self-Defined Time Zone
of the Prime Meridian (A.K.A: International or Greenwich Meridian); If it
is negative (-), the local time zone is east.
M4.1.0,M11.1.0
The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec).
The 2nd number indicates the nth iteration of the weekday: (1st Sunday,
3rd Tuesday…). Normally 1, 2, 3, 4 are used. If 5 is used, it means the
last iteration of the weekday.
The 3rd number indicates weekday: 0,1,2,..,6 ( for Sun, Mon,
Tues, ... ,Sat).
Therefore, this example is the DST which starts from the First Sunday of
April to the 1st Sunday of November.
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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 19: Set Time Manually
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->Time 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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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, assignment and provisioning. The
UCM6100 creates XML config file to the detected/assigned Grandstream device and accomplishes the
following configurations on the device after the provisioning:

A UCM6100 extension will be assigned and registered on the phone.

SIP-related network settings such as "NAT traversal" and "Use Random Port" are configured on the
phone.

Call feature settings such as "Public Mode", "Voicemail User ID", "Dial Plan" and "Auto Answer".

LDAP client configurations will be set up automatically on the phone to use the default LDAP directory
generated in the UCM6100 LDAP server.

Date format, time format and time zone settings for the phone to be provisioned.
This section explains how zero config works on the UCM6100. The settings for this feature can be
accessed via Web GUI->PBX->Basic/Call Routes->Zero Config.
AUTO PROVISIONING
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 20: 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 reboot
again to 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, http://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->Basic/Call Routes->Zero Config, click on
"Auto Provision Settings" and fill in the auto provision information.
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Figure 21: Auto Provision Settings
Table 21: Auto Provision Settings
Enable Zero Config
Enable or disable the zero config feature on the PBX. The default setting
is disabled.
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.
Zero Config Extension
Segment
Click on the link "Zero Config Extension Segment" to specify the
extension range to be assigned if "Automatically Assign Extension" is
enabled. The default range is 5000-6299.
If enabled, the extension list will be sent out to the device after receiving
Enable Pick Extension
the device's request. This feature is for the GXP series phones that
support selecting extension to be provisioned via phone's LCD. The
default setting is disabled.
Pick Extension Segment
Pick Extension Period (hour):
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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.
Specify the number of minutes to allow the phones being provisioned to
pick extensions.
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Provision with Date and Time
If enabled, the end devices will be provisioned with the date format, time
Settings
format and time zone as configured in below options.
Date Format
Select data format for the end device to be provisioned.
Time Format
Time Zone
Select time format (12-hour clock or 24-hour clock) for the end device to
be provisioned.
Select time zone for the end device to be provisioned.
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.
MANUAL PROVISIONING
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->Basic/Call Routes->Zero
Config->Auto Provisioning Settings.
Figure 22: Auto Discover
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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 23: Discovered Devices
ASSIGNMENT
In the discovered list, click on
to open the edit dialog to assign an extension or multiple extensions to
this device. Hot-Desking can also be enabled from this edit page.
Figure 24: Assign Extension To Device
After saving the edit dialog, the XML config file will be generated in the UCM6100. Reboot the phone or
trigger the phone to download the config file by clicking on
list.
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icon for the entry in the zero config device
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CREATE NEW DEVICE
Users could also directly create a new device and assign the extension before the device is discovered by
the UCM6100. Once the device is plugged in, it can then be discovered and provisioned by the UCM6100.
Click on "Create New Device" and the following dialog will show. Enabled Hot-Desking(Optional), fill in the
MAC address (required), IP address (optional), Version (optional), Model (optional) and the extension to
assign to the device. Click on "Save" to add the device to the provision list.
Figure 25: Create New Device
PROVISIONING
After the successful discovery and assignment configuration on the UCM6100, the device will start
downloading the config file and take the new configuration with the extension registered.
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EXTENSIONS
CREATE NEW USER
CREATE NEW SIP EXTENSION
To manually create new SIP user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on
"Create New User"->"Create New SIP Extension" and a new dialog window will show for users to fill in the
extension information. The configuration parameters are as follows.
Table 22: SIP Extension Configuration Parameters
General
Extension
The extension number associated with the user.
Configure the CallerID Number that would be applied for outbound calls
from this user.
CallerID Number
Note:
The ability to manipulate your outbound Caller ID may be limited by your
VoIP provider.
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Permission
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege 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.
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.
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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.
Configure the Call Forward Busy target number. If not configured, the
Call Forward Busy
Call Forward Busy feature is deactivated. The default setting is
deactivated.
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
Auto Record
default setting is disabled. The recording files can be accessed under
web GUI->CDR->Recording Files.
Skip Voicemail Password
Verification
Support Hot-Desking Mode
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 enabled, SIP Password will accept only alphabet characters and digits;
AuthID will be changed to the same as Extension.
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
Language
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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
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under web GUI->PBX->Internal Options->Language. The dropdown
list 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.
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
Can Reinvite
with the 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
Keep-alive Frequency
Auth ID
If enabled, empty SDP packet will be sent to the SIP server periodically
to keep the NAT port open. The default setting is "Yes".
Configure the Keep-alive interval (in seconds) to check if the host is up.
The default setting is 60 seconds.
Configure the authentication ID for the user. If not configured, the
extension number will be used for authentication.
If the end device/phone has an assigned PSTN telephone number, this
TEL URI
field should be set to "User=Phone". Then a "User=Phone" parameter
will be attached to the Request-Line and TO header in the SIP request
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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 disabled.
Enable to detect Fax signal from the user/trunk during the call and send
the received Fax to the Email address configured for this extension. If no
Email address can be found for the user, send the received Fax to the
Fax Detection
default
Email
address
in
Fax
setting
page
under
web
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
Skip Trunk Auth
Dial Trunk Password
If enabled, users will not need enter the "PIN Set" required by the
outbound rule to make outbound calls. The default setting is "No".
Configure personal password when making outbound calls via trunk.
This option controls how the extension can be used on devices within
different types of network.

Allow All
Device in any network can register this extension.

Local Subnet Only
Only the user in specific subnet can register this extension. Up to
Strategy
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
Codec Preference
are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729,
G.723, ILBC, ADPCM, H.264, H.263 and H.263p.
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.
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Table 23: IAX Extension Configuration Parameters
General
Extension
The extension number associated with the user.
Configure the CallerID Number that would be applied for outbound calls
from this user.
CallerID Number
Note:
The ability to manipulate your outbound Caller ID may be limited by your
VoIP provider.
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Permission
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege 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.
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.
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.
Configure the Call Forward Busy target number. If not configured, the
Call Forward Busy
Call Forward Busy feature is deactivated. The default setting is
deactivated.
Configure the number of seconds to ring the user before the call is
forwarded to voicemail (voicemail is enabled) or hang up (voicemail is
Ring Timeout
disabled). If not specified, the default ring timeout is 60 seconds on the
UCM6100, which can be configured in the global ring timeout setting
under web GUI->Internal Options->IVR Prompt: General Preference.
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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
Auto Record
default setting is disabled. The recording files can be accessed under
web GUI->CDR->Recording Files.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is
skipped. If enabled, this would allow one-button voicemail access. By
default this option is disabled.
If selected, this IAX 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
Language
list 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.
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
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SRTP
Enable SRTP for the call. The default setting is disabled.
Enable to detect Fax signal from the user/trunk during the call and send
the received Fax to the Email address configured for this extension. If no
Email address can be found for the user, send the received Fax to the
Fax Detection
default
Email
address
in
Fax
setting
page
under
web
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
Skip Trunk Auth
Dial Trunk Password
If enabled, users will not need enter the "PIN Set" required by the
outbound rule to make outbound calls. The default setting is "No".
Configure personal password when making outbound calls via trunk.
This option controls how the extension can be used on devices within
different types of network.

Allow All
Device in any network can register this extension.

Local Subnet Only
Only the user in specific subnet can register this extension. Up to
Strategy
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
Codec Preference
are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729,
G.723, ILBC, ADPCM, H.264, H.263 and H.263p.
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 24: FXS Extension Configuration Parameters
General
Extension
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The extension number associated with the user.
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Analog Station
Select the FXS port to be assigned for this extension.
Configure the CallerID Number that would be applied for outbound calls
from this user.
CallerID Number
Note:
The ability to manipulate your outbound Caller ID may be limited by your
VoIP provider.
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Permission
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege 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.
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.
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.
Configure the Call Forward Busy target number. If not configured, the
Call Forward Busy
Call Forward Busy feature is deactivated. The default setting is
deactivated.
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.
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Enable automatic recording for the calls using this extension. The
Auto Record
default setting is disabled. The recording files can be accessed under
web GUI->CDR->Recording Files.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is
skipped. If enabled, this would allow one-button voicemail access. By
default this option is disabled.
If selected, this FXS 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
Language
list 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.
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
MIN RX Flash
hook-flash must remain unpressed for the PBX to consider the event as
a valid flash event. The valid range is 30ms to 1000ms. The default
setting is 200ms.
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Configure the maximum period of time (in milliseconds) that the
MAX RX Flash
hook-flash must remain unpressed for the PBX to consider the event as
a valid flash event. The minimum period of time is 256ms and it can't be
modified. The default setting is 1250ms.
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. The default setting is
1.
Other Settings
Enable to detect Fax signal from the user/trunk during the call and send
the received Fax to the Email address configured for this extension. If no
Email address can be found for the user, send the received Fax to the
Fax Detection
default
Email
address
in
Fax
setting
page
under
web
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
Skip Trunk Auth
Dial Trunk Password
If enabled, users will not need enter the "PIN Set" required by the
outbound rule to make outbound calls. The default setting is "No".
Configure personal password when making outbound calls via trunk.
BATCH ADD EXTENSIONS
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BATCH ADD SIP EXTENSIONS
Under Web GUI->PBX->Basic/Call Routes->Extensions, click on "Batch Add Extensions"->"Batch Add
SIP Extensions".
Table 25: Batch Add SIP Extension Parameters
General
Start Extension
Create Number
Configure the starting extension number of the batch of extensions to be
added.
Specify the number of extensions to be added. The default setting is 5.
Assign permission level to the user. The available permissions are
"Internal", "Local", "National" and "International" from the lowest level to
the highest level. The default setting is "Internal".
Permission
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege in order to make outbound calls 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
Ring Timeout
UCM6100, which can be configured in the global ring timeout setting
under web GUI->Internal Options->IVR Prompt: General Preference.
The valid range is between 5 seconds and 600 seconds.
Note:
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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
Auto Record
default setting is disabled. The recording files can be accessed under
web GUI->CDR->Recording Files.
Skip Voicemail Password
Verification
When user dials voicemail code, the password verification IVR is
skipped. If enabled, this would allow one-button voicemail access. By
default this option is disabled.
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
TEL URI
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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".
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
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will be attached to the Request-Line and TO header in the SIP request
to indicate the E.164 number. If set to "Enable", "Tel:" will be used
instead of "SIP:" in the SIP request. The default setting is disabled.
Other Settings
SRTP
Enable SRTP for the call. The default setting is "No".
Enable to detect Fax signal from the user/trunk during the call and send
the received Fax to the Email address configured for this extension. If no
Email address can be found for the user, send the received Fax to the
Fax Detection
default
Email
address
in
Fax
setting
page
under
web
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
This option controls how the extension can be used on devices within
different types of network.

Allow All
Device in any network can register this extension.

Local Subnet Only
Only the user in specific subnet can register this extension. Up to
Strategy
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
Codec Preference
are: 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".
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Table 26: 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
the highest level. The default setting is "Internal".
Permission
Note:
Users need to have the same level as or higher level than an outbound
rule's privilege in order to make outbound calls 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
Auto Record
default setting is disabled. The recording files can be accessed under
web GUI->CDR->Recording Files.
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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.
IAX Settings
Max Number of Calls
Configure the maximum number of calls allowed for each remote IP
address.
Configure to enable/disable requiring call token. If set to "Auto", it might
Require Call Token
lock out users who depend on backward compatibility when peer
authentication credentials are shared between physical endpoints. The
default setting is "Yes".
Other Settings
SRTP
Enable SRTP for the call. The default setting is "No".
Enable to detect Fax signal from the user/trunk during the call and send
the received Fax to the Email address configured for this extension. If no
Email address can be found for the user, send the received Fax to the
Fax Detection
default
Email
address
in
Fax
setting
page
under
web
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
This option controls how the extension can be used on devices within
different types of network.

Allow All
Device in any network can register this extension.

Local Subnet Only
Only the user in specific subnet can register this extension. Up to
Strategy
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
Codec Preference
are: PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, ILBC,
ADPCM, LPC10, H.264, H.263 and H.263p.
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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)
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->Basic/Call Routes->Zero Config->Auto Provisioning Settings.

Delete single extension
Click on
to delete the extension. Or select the checkbox of the extension and then click on "Delete
Selected Extensions".

Modify selected extensions
Select the checkbox for the extension(s). Then click on "Modify Selected Extensions" to edit the
extensions in a batch.

Delete selected extensions
Select the checkbox for the extension(s). Then click on "Delete Selected Extensions" to delete the
extension(s).
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EXPORT EXTENSIONS
The extensions configured on the 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 26: Export Extensions
The exported csv file can be serve as a template for users to fill in desired extension information to be
imported to the 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 27: Import Extensions
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4. Select the option in "On Duplicate Extension" to define how the duplicate extension(s) in the imported
csv file should be treated by the PBX.

Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension
information as previously configured without change.

Delete and Recreate: The current extension previously configured will be deleted and the
duplicate extension in the csv file will be loaded to the PBX.

Update Information: The current extension previously configured in the PBX will be kept. However,
if the duplicate extension in the csv file has different configuration for any options, it will override
the configuration for those options in the extension.
5. Click on
to select csv file from local directory in the PC.
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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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 27: 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.
Advanced Options
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 "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)
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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.
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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".
Enable to detect Fax signal from the trunk during the call and send the
received Fax to the default Email address in Fax setting page under web
Fax Detection
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
Caller ID Scheme
Select the Caller ID scheme for this trunk. The default setting is
"Bellcore/Telcordia".
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
If selected, the trunk will be disabled.
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)".
Busy Tone
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Syntax:
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f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];
Frequencies are in Hz and cadence on and off are in ms.
Frequencies Range: [0, 4000)
Busy Level Range: (-300, 0)
Cadence Range: [0, 16383].
Select Tone Country "Custom" to manually configure Busy Tone value.
Default value:
[email protected],[email protected],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:
[email protected],[email protected],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 28: UCM6100 FXO Tone Settings
4. Click on "Detect" to start PSTN detection.
Figure 29: 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 30: UCM6100 PSTN Detection: Auto Detect

If there is only one FXO port connected to PSTN line, use the following settings for auto-detection.
Figure 31: 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.
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Table 28: PSTN Detection For Analog Trunk
Select "Auto Detect" or "Semi-auto Detect" for PSTN detection.

Auto Detect
Please make sure two or more channels are connected to the
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.
Detect Model

Semi-auto Detect
Semi-auto detection requires answering or hanging up the call
manually. Please make sure one channel is connected to the
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.
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
Click on "Create New SIP Trunk" or "Create New IAX Trunk" to add a new VoIP trunk.

Click on

Click on
to configure Direct Outward Dialing (DOD) for the SIP Trunk.

Click on
to start LDAP Sync.

Click on
to configure detailed parameters for the VoIP trunk.
to delete the VoIP trunk.
The VoIP trunk options are listed in the table below.
Table 29: SIP Trunk Configuration Parameters
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
TEL URI
attached to the Request-Line and TO header in the SIP request to
indicate the E.164 number. If set to "Enable", "Tel:" will be used instead
of "SIP:" in the SIP request. The default setting is disabled.
Need Registration
Username
Password
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.
Auth ID
Enter the Authentication ID for "Register SIP Trunk" type.
Outbound Proxy
Enter the IP address or URL of the outbound proxy for "Register SIP
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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.
Peer SIP Trunk Configuration Parameters
Provider Name
Host Name
Configure the provider name for the VoIP trunk. This is a unique label to
identify the trunk when listed in outbound rules, inbound rules and etc.
Configure the IP address or URL for the VoIP provider 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
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
TEL URI
attached to the Request-Line and TO header in the SIP request to
indicate the E.164 number. If set to "Enable", "Tel:" will be used instead
of "SIP:" in the SIP request. The default setting is disabled.
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.
Caller ID
When making outgoing calls, the following rules are used to determine
which CallerID will be used if they exist:
Firmware Version 1.0.6.10

The CallerID configured for the extension will be looked up first.

If no CallerID configured for the extension, the CallerID configured
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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.
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 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.
DID Mode
Enable Qualify
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".
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
Qualify Timeout
ms) 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.
The Maximum Number of Call
The maximum number of concurrent calls using the trunk. The default
Lines
settings 0, which means unlimited.
Enable to detect Fax signal from the trunk during the call and send the
received Fax to the default Email address in Fax setting page under web
Fax Detection
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
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
Sync LDAP Password
decryption during the LDAP sync process. The password must be the
same on both UCM6100 peers o ensure successful synchronization.
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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.
Register SIP Trunk Configuration Parameters
Provider Name
Host Name
Configure the provider name for the VoIP trunk. This is a unique label to
identify the trunk when listed in outbound rules, inbound rules and etc.
Configure the IP address or URL for the VoIP provider 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
When enabled, it can avoid overridden by extension's CID if the
extension has CID configured. The default setting is enabled.
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
TEL URI
attached to the Request-Line and TO header in the SIP request to
indicate the E.164 number. If set to "Enable", "Tel:" will be used instead
of "SIP:" in the SIP request. The default setting is disabled.
Need Registration
Configure whether to register the trunk to the external server or not. The
default setting is Yes.
Username
Enter the username to register to the trunk from the provider.
Password
Enter the password to register to the trunk from the provider.
This is the authentication ID for the UCM6100 to register to the trunk if
Auth ID
required by the provider. If not specified, the CallerID name will be sued
for authentication.
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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 can be used to override the From Header.
From Domain
For example, "trunk.UCM6100.provider.com" is the From Domain in
From Header: sip:[email protected]
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
From User
for registration (single trunk) with multiple DIDs.
For example, "1234567" is the From User in From Header:
sip:[email protected]
Outbound Proxy Support
Outbound Proxy
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.
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.
DID Mode
Enable Qualify
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".
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
Qualify Timeout
ms) 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.
The Maximum Number of Call
The maximum number of concurrent calls using the trunk. The default
Lines
settings 0, which means unlimited.
Enable to detect Fax signal from the trunk during the call and send the
received Fax to the default Email address in Fax setting page under web
Fax Detection
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
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SRTP
Enable SRTP for the VoIP trunk. The default setting is "No".
Table 30: IAX Trunk Configuration Parameters
Create New IAX Trunk
Select the VoIP trunk type.
Type
Provider Name
Host Name
Keep Trunk CID
Username
Password
Disable This Trunk

Peer IAX Trunk

Register IAX Trunk
Configure a unique label to identify this trunk when listed in outbound
rules, inbound rules and etc.
Configure the IP address or URL for the VoIP provider’s server of the
trunk.
If enabled, the trunk CID will not be overridden by extension's CID when
the extension has CID configured. The default setting is "No".
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.
Peer IAX Trunk Configuration Parameters
Provider Name
Host Name
Keep Trunk CID
Disable This Trunk
Configure the provider name for the VoIP trunk. This is a unique label to
identify the trunk when listed in outbound rules, inbound rules and etc.
Configure the IP address or URL for the VoIP provider 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.
Caller ID
When making outgoing calls, the following rules are used to determine
which CallerID will be used if they exist:
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
The CallerID configured for the extension will be looked up first.

If no CallerID configured for the extension, the CallerID configured
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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.
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
Qualify Timeout
ms) 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.
The Maximum Number of Call
The maximum number of concurrent calls using the trunk. The default
Lines
settings 0, which means unlimited.
Enable to detect Fax signal from the trunk during the call and send the
received Fax to the default Email address in Fax setting page under web
Fax Detection
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
Register IAX Trunk Configuration Parameters
Provider Name
Host Name
Keep Trunk CID
Disable This Trunk
Configure the provider name for the VoIP trunk. This is a unique label to
identify the trunk when listed in outbound rules, inbound rules and etc.
Configure the IP address or URL for the VoIP provider server of the
trunk.
When enabled, it can avoid overridden by extension's CID if the
extension has CID configured. The default setting is enabled.
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
Caller ID
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
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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.
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
Qualify Timeout
ms) 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.
The Maximum Number of Call
The maximum number of concurrent calls using the trunk. The default
Lines
settings 0, which means unlimited.
Enable to detect Fax signal from the trunk during the call and send the
received Fax to the default Email address in Fax setting page under web
Fax Detection
GUI->PBX->Internal Options->Fax/T.38.
Note:
If enabled, Fax Pass-through cannot be used.
Direct Outward Dialing (DOD)
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.
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Example of how DOD is used:
Company ABC has a SIP trunk. This SIP trunk has 4 DIDs associated to it. The main number of the office
is routed to an auto attendant. The other three numbers are direct lines to specific users of the company.
At the moment when a user makes an outbound call their caller ID shows up as the main office number.
This poses a problem as the CEO would like their calls to come from their direct line. This can be
accomplished by configuring DOD for the CEO’s extension.
Steps on how to configure DOD on the UCM:
1. To setup DOD go to 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 32: 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 33: Edit DOD
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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.

Click on
to move the outbound route up/down to arrange the priority of the outbound
rule. The outbound rule listed on the top has higher priority. When the dialing pattern matches two or
more outbound rules (for example, the same pattern is configured for 2 different trunks; or dialing out
1000 matches pattern 1xxx for trunk 1 and pattern 100x for trunk 2), the one listed on the top will be
used.
Table 31: Outbound Route Configuration Parameters
Calling Rule Name
Configure the name of the calling rule (e.g., local, long_distance, and
etc). Letters, digits, _ and - are allowed.

All patterns are prefixed with the "_".

Special characters:
X: Any Digit from 0-9.
Z: Any Digit from 1-9.
Pattern
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
Privilege Level
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Configure the password for users to use this rule when making outbound
calls.
Select privilege level for the outbound rule.

Internal: The lowest level required. All users can use this rule.
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
Local: Users with Local, National, or International level are allowed
to use this rule.

National: Users with National or International level are allowed to
use this rule.

International:
The
highest
level
required.
Only
users
with
international level can use this rule.
The default setting is "Disable". Please be aware of the potential security
risks when using "Internal" level, which means all users can use this
outbound rule to dial out from the trunk.
When enabled, users could specify extensions allowed to use this
outbound route. "Privilege Level" is automatically disabled if using
"Enable Filter on Source Caller ID".
The following two methods can be used at the same time to define the
extensions as the source caller ID.
1. Select available extensions/extension groups from the left to the
right. This allows users to specify arbitrary single extensions
Enable Filter on Source Caller
ID
available in the PBX.
2. Custom Dynamic Route: define the pattern for the source caller ID.
This allows users to define extension range instead of selecting
them one by one.

All patterns are prefixed with the "_".

Special characters:
X: Any Digit from 0-9.
Z: Any Digit from 1-9.
N: Any Digit from 2-9.
".": Wildcard. Match one or more characters.
"!": Wildcard. Match zero or more characters immediately.
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
Strip
selected trunk.
Example:
The users will dial 9 as the first digit of a long distance calls. However, 9
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should not be sent out via analog lines and the PSTN line. In this case, 1
digit should be stripped before the call is placed.
Specify the digits to be prepended before the call is placed via the trunk.
Prepend
Those digits will be prepended after the dialing number is stripped.
Use Failover Trunk
Failover trunks can be used to make sure that a call goes through an
alternate route, when the primary trunk is busy or down. If "Use Failover
Trunk" is enabled and "Failover trunk" is defined, the calls that cannot be
placed via the regular trunk may have a secondary trunk to go through.
Failover Trunk
Example:
The user's primary trunk is a VoIP trunk and the user would like to use
the PSTN when the VoIP trunk is not available. The PSTN trunk can be
configured as the failover trunk of the VoIP trunk.
Allows the user to specify the number of digits that will be stripped from
the beginning of the dialed string before the call is placed via the
selected trunk.
Strip
Example:
The users will dial 9 as the first digit of a long distance calls. However, 9
should not be sent out via analog lines and the PSTN line. In this case, 1
digit should be stripped before the call is placed.
Specify the digits to be prepended before the call is placed via the trunk.
Prepend
Those digits will be prepended after the dialing number is stripped.
INBOUND ROUTES
Inbound routes can be configured via Web GUI->PBX->Basic/Call Routes->Inbound Routes.

Click on "Create New Inbound Rule" to add a new inbound route.

Click on "Blacklist" to configure blacklist for all inbound routes.

Click on
to edit the inbound route.

Click on
to delete the inbound route.
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INBOUND RULE CONFIGURATIONS
Table 32: 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.
Select privilege level for the inbound rule when a VoIP trunk is selected
in "Trunks" field.

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.

Privilege Level
National: Users with National or International level are allowed to
use this rule.

International:
The
highest
level
required.
Only
users
with
international level can use this rule.
This setting is used to compare with the outbound trunk's permission
level when the inbound call dials out via a trunk on the UCM6100.
Therefore, it's usually used only when the "Default Destination" is set to
"By DID".
Select the default destination for the inbound call.
Default Destination
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
Extension

Voicemail

Conference Room

Queue

Ring Group

Page/Intercom
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
Voicemail Group

Fax

DISA

IVR

Dial By Name

By DID
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
This option shows up when "By DID" is selected. It configures the
number of digits to be stripped from the beginning of the DID number.
This option shows up when "By DID" is selected. If enabled, the trunk
name will be prepended to the display name.
This option shows up when "By DID" is selected. If enabled, external
Dial Trunk
users calling in using "By DID" are allowed to dial out via the PBX
internal trunks.
This option shows up when "By DID" is selected. Users can select the
DID destination for the external users to reach. Only the selected
category can be reached by DID using this inbound route.
DID Destination

Extension

Conference

Call Queue

Ring Group

Paging/Intercom Groups

IVR

Voicemail Groups

Fax Extension

Dial By Name
Time Condition
Start Time
Select the start time "hour:minute" for the trunk to use the inbound rule.
End Time
Select the end time "hour:minute" for the trunk to use the inbound rule.
Date
Select "By Week" or "By Day" and specify the date for the trunk to use
the inbound rule.
Week
Select the day in the week to use the inbound rule.
Destination
Select the default destination for the inbound call.
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
By DID (for VoIP trunk only)
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" under "DID Features" dialog. If the dialed number
matches the DID pattern, the call will be allowed to go through.

Extension

Voicemail

Conference Room

Queue

Ring Group

Page

Voicemail Group

Fax

DISA

IVR

Dial By Name
BLACKLIST CONFIGURATIONS
In the UCM6100, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature,
manage the Blacklist by clicking on "Blacklist".
Figure 34: Blacklist Configuration Parameters
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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.
.
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.

UCM6108/6116 supports up to 6 conference bridges allowing up to 32 simultaneous PSTN or IP
participants.
The
conference
bridge
configurations
can
be
accessed
under
Web
GUI->PBX->Call
Features->Conference. In this page, users could create, edit, view, invite, manage the participants and
delete conference bridges. The conference bridge status and conference call recordings (if recording is
enabled) will be displayed in this web page as well.
CONFERENCE BRIDGE CONFIGURATIONS

Click on "Create New Conference Room" to add a new conference bridge.

Click on
to edit the conference bridge.

Click on
to delete the conference bridge.
Table 33: 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
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conference call.
Note:

If "Public Mode" is enabled, the password is not required to join the
conference bridge thus this field is invalid.

Enable Caller Menu
The password has to be at least 4 characters.
If enabled, conference participant could press the * key to access the
conference bridge menu. The default setting is "No".
If enabled, the calls in this conference bridge will be recorded
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 For First
If enabled, the UCM6100 will play Hold music to the first participant in
Caller
the conference until another user joins in. 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 35: 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 34: 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 download the
to delete the recording.
Figure 36: 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 35: IVR Configuration Parameters
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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After playing the prompts in the IVR, the UCM6100 will wait for the
DTMF entry within the timeout (in seconds). If no DTMF entry is
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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.
Select the event for each key pressing for 0-9, *, Timeout and Invalid.
The event options are:
Key Press Event
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
Extension

Voicemail

Conference Rooms

Voicemail Group

IVR

Ring Group

Queues

Page Group

Fax

IVR Prompt

Hangup

DISA

Dial By Name
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CREATE IVR PROMPT
To record new IVR prompt or upload IVR prompt to be used in IVR, click on “Prompt” next to the “Welcome
Prompt” option and the users will be redirected to IVR Prompt page. Or users could go to Web
GUI->PBX->Internal Options->IVR Prompt page directly.
Figure 37: 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 IVR PROMPT
In the UCM6100 Web GUI->PBX->Internal Options->IVR Prompt page, click on “Record New IVR
Prompt” and follow the steps below to record new IVR prompt.
Figure 38: Record New IVR Prompt

Specify the IVR file name.

Select the format (GSM or WAV) for the IVR prompt file to be recorded.

Select the extension to receive the call from the UCM6100 to record the IVR prompt.
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
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 IVR PROMPT
If the user has a pre-recorded IVR prompt file, click on “Upload IVR Prompt” in Web GUI->PBX->Internal
Options->IVR Prompt page to upload the file to the 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 39: Upload IVR Prompt
Click on
to select audio file from local PC and click on
to start uploading. Once uploaded, the file
will appear in the IVR Prompt web page.
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LANGUAGE SETTINGS FOR VOICE PROMPT
The UCM6100 supports multiple languages in web GUI as well as system voice prompt. The following
languages are currently supported in system voice prompt:
English (United States)
Arabic
Chinese
Dutch
English (United Kingdom)
French
German
Greek
Hebrew
Italian
Polish
Portuguese
Russian
Spanish
Swedish
Turkish
English (United States) and Chinese voice prompts are built in with the 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.
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Figure 40: Language Settings For Voice Prompt
A new dialog window of voice prompt package list will be displayed. Users can see the version number
(latest version available V.S. current installed version), package size and options to upgrade or download
the language.
Figure 41: Voice Prompt Package List
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Click on
to download the language to the UCM6100. The installation will be automatically started
once the downloading is finished.
Figure 42: 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.
CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE
The UCM6100 provides interface from web GUI for users to customize their own voice prompts. Users
could directly upload the package from web GUI. For detailed instructions on voice prompt customizing
and uploading, please refer to the link below:
http://www.grandstream.com/products/ucm_series/UCM6100/documents/UCM6100_voiceprompt_custom
ization.zip
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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 36: 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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announced at the beginning of the voicemail message. The default
setting is "No".
Announce Message Duration
If enabled, the message duration will be announced at the beginning of
the voicemail message. The default setting is "No".
If enabled, a brief introduction (received time, received from, and etc) of
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".
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.
Figure 43: Voicemail Email Settings
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Table 37: 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

\t: TAB

${VM_NAME}: Recipient's first name and last name

${VM_DUR}: The duration of the voicemail message

${VM_MAILBOX}: The recipient's extension

${VM_CALLERID}: The caller ID of the person who has left the
message

${VM_MSGNUM}: The number of messages in the mailbox

${VM_DATE}: The date and time when the message is left
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.
Figure 44: Voicemail Group
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Table 38: 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 45: 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 39: 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
Ring Strategy
will stop ringing.

Ring in order.
Ring the members with the order configured in ring group list. If the
first member doesn’t answer the call, it will stop ringing the first
member and start ringing the second member.
Configure the number of seconds to ring each member. If set to 0, it will
Ring Timeout on Each Member
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keep ringing. The default setting is 30 seconds.
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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 46: 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 47: Paging/Intercom Group
Table 40: 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".
Page/Intercom Group
Select available users from the left side to the paging/intercom group
Members
member list on the right.

Click on
to edit the paging/intercom group.
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
Click on

Click on "Paging/Intercom Group Settings" to edit Alert-Info Header. This header will be included in the
to delete the paging/intercom group.
SIP INVITE message sent to the callee in paging/intercom call.
Figure 48: 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.

Click on "Create New Queue" to add call queue.
Figure 49: Call Queue

Click on
to edit the call queue. The call queue configuration parameters are listed in the table
below.
Table 41: Call Queue Configuration Parameters
Extension
Configure the call queue extension.
Name
Configure the call queue name to identify the call queue.
Select the strategy for the call queue.

Ring All
Ring all available Agents simultaneously until one answers.

Linear
Ring agents in the specified order.

Strategy
Least Recent
Ring the agent who has been called the least recently.

Fewest Calls
Ring the agent with the fewest completed calls.

Random
Ring a random agent.

Round Robin
Ring the agents in Round Robin scheduling with memory.
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The default setting is "Ring All".
Select the Music On Hold class for the call queue.
Music On Hold
Note:
Music On Hold classes can be managed from Web GUI->
PBX->Internal Options->Music On Hold.
Configure whether the callers will be disconnected from the queue or not
if the queue has no agent anymore. The default setting is "Strict".

Yes
Callers will be disconnected from the queue if all agents are paused
or 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.
Dynamic Login Password
Ring Time Out
If enabled, the configured PIN number is required for dynamic agent to
log in. The default setting is disabled.
Configure the number of seconds an agent will ring before the call goes
to the next agent. The default setting is 15 seconds.
Configure the number of seconds before a new call can ring the queue
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.
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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.
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.
Figure 50: 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.
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
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".
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EXTENSION GROUPS
The UCM6100 extension group feature is added since firmware version 1.0.5.14. Users could assign
extensions to 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 51: Edit Extension Group

Click on
to delete the extension group.
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USE EXTENSION GROUPS
Here is an example where the extension group can be used. Go to Web GUI->PBX->Basic/Call
Routes->Outbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and
extension groups will show up for users to select.
Figure 52: 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 53: Edit Pickup Group

Click on
to delete the pickup group.
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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 54: Music On Hold Default Class

Click on "Create New MOH Class" to add a new Music On Hold class.

Click on
to configure the MOH class sort method to be "Alpha" or "Random" for the sound files.

Click on
next to the selected Music On Hold class to delete this Music On Hold class.

Click on
to select music file from local PC and click on
to start uploading. The music file
uploaded has to be 8 KHz Mono format with size smaller than 5M.

Click on
next to the sound file to delete it from the selected Music On Hold Class.
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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 42: 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.
Maximum Transfer Rate
The possible values are 2400, 4800, 7200, 9600, 12000 and 14400. The
default setting is 14400.
Configure the minimum transfer rate during the Fax rate negotiation. The
Minimum Transfer Rate
possible values are 2400, 4800, 7200, 9600, 12000 and 14000. The
default setting is 2400.
Configure the 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.
Template Variables
Firmware Version 1.0.6.10
The template variables are:

${CALLERIDNUM} : Caller ID Number

${CALLERIDNAME} : Caller ID Name

${RECEIVEEXTEN} : The extension to receive the Fax

${FAXPAGES} : Number of pages in the Fax
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
${VM_DATE} : The date and time when the Fax is received

Click on
to edit the Fax extension.

Click on
to delete the Fax extension.
SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE
The following instructions 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 Detection: Make sure "Fax Detection" option is set to "No".
Figure 55: 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 56: 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 57: 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 58: 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 59: Inbound Route To Fax Extension
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5. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number
will be converted to PDF file and sent to the Email address [email protected] as
attachment.
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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 60: Create New DISA
Table 43: 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
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route's 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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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 is normally used
to park the call by configuring the MPK number as call park feature code (e.g., 700). Users could also
use "Call Park" mode to monitor and pick up the call on this parking lot by configuring the MPK number
as parking lot number (e.g., 701).
EVENT LIST
Besides BLF, users can also configure the phones to monitor event list. 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.
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Figure 61: Create New Event List
Table 44: 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 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
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
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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 62: 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 63: Dial By Name Group In IVR Key Pressing Events
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Figure 64: 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 65: 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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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 45: 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
complete the attended transfer.
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
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callee.

Default code: #72.

Enter the code during active call to park the call.

Options
Disable
Call Park
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

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 Code Digits Timeout

Default Setting: 1000.
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
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.
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.
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
Default Code: *8.

This code is for the pickup group which can be assigned for
each extension on the extension configuration page.
Pickup Extension

If there is an incoming call to an extension, the other
extensions within the same pickup group can dial *8 directly to
pick up the call.

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 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.
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 66: Download Recording File From CDR Page
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The above recorded call's recording files are also listed under the UCM6100 web GUI->CDR->Recording
Files.
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.

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 46: 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
Operator Extension
0 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:
1. Password for voicemail, voicemail group, outbound route, DISA, call
queue and conference requires non-repetitive and non-sequential
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digits, with a minimum length of 4 digits. Repetitive digits pattern
(such as 0000, 1111, 1234, 2345, and etc), or common digits pattern
(such as 111222, 321321 and etc) are not allowed to be configured
as password.
2. Password for extension registration, web GUI admin login, LDAP
and LDAP sync requires alphanumeric characters containing at
least two categories of the following, with a minimum length of 4
characters.

Numeric digits

Lowercase alphabet characters

Uppercase alphabet characters

Special characters
If enabled, random password will be generated when the extension is
Enable Random Password
created. The default setting is "Yes". It is recommended to enable it for
security purpose.
If
set
to
"Yes",
users
could
disable
the
extension
range
pre-configured/configured on the UCM6100. The default setting is "No".
The default extension range assignment is:
Disable Extension Range

User Extensions: 1000-6299

Pick Extensions: 4000-4999

Auto Provision Extensions: 5000-6299

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
Note:
It is recommended to keep the system assignment to avoid
inappropriate usage and unnecessary issues.
INTERNAL OPTIONS/JITTER BUFFER
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Table 47: 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 48: Internal Options/RTP Settings
RTP Start
Configure the RTP port starting number. The default setting is 10000.
RTP End
Configure the RTP port ending address. The default setting is 20000.
Configure to enable or disable strict RTP protection. If enabled, RTP
Strict RTP
packets that do not come from the source of the RTP stream will be
dropped. The default setting is "Disable".
RTP Checksums
Configure to enable or disable RTP Checksums on RTP traffic. The
default setting is "Disable".
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.
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Figure 67: 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 68: FXO Ports ACIM Settings
Table 49: 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
Select country to set the On Hook Speed, Ringer Impedance, Ringer
Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum
FXO Opermode
Operational Loop Current, and AC Impedance as predefined for your
country's analog line characteristics. The default setting is "United
States of America (USA)".
Select country to set the On Hook Speed, Ringer Impedance, Ringer
Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum
FXS Opermode
Operational Loop Current, and AC Impedance as predefined for your
country's analog line characteristics. The default setting is "United
States of America (USA)".
Configure to enable or disable override Two-Wire Impedance Synthesis
FXS TISS Override
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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
Table 50: 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
Firmware Version 1.0.6.10
Configure the number of seconds between STUN Refreshes. The
default setting is 30 seconds.
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INTERNAL OPTIONS/PAYLOAD
The UCM6100 payload type for audio codecs and video codes can be configured here.
Table 51: Internal Options/Payload
AAL2-G.726
DTMF
G.721 Compatible
G.726
Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword
packing). The default setting is 112.
Configured payload type for DTMF. The default setting is 101.
Configure to enable/disable G.721 compatible. The default setting is
Yes.
Configure the payload type for G.726 if "G.721 Compatible" is disabled.
The 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 52: 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 53: IAX Settings/Registration
IAX Registration Options
Min Reg Expire
Max Reg Expire
Firmware Version 1.0.6.10
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.
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IAX Thread Count
IAX Max Thread Count
Configure the number of IAX helper threads. The default setting is 10.
Configure the maximum number of IAX threads allowed. The default
setting is 100.
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 54: IAX Settings/Static Defense
Enter a single IP address or a range of IP addresses for which call token
validation is not required.
Call Token Optional
For example:
11.11.11.11
11.11.11.11/22.22.22.22.
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Max Call Numbers
Max Unvalidated Call Numbers
Call Number Limits
Configure the maximum number of calls allowed for a single IP address.
Configure the maximum number of unvalidated calls for all IP
addresses.
Configure to limit the number of calls for a give IP address of IP range.
Enter the IP address or a range of IP addresses to be considered for call
number limits.
IP or IP Range
For example:
11.11.11.11
11.11.11.11/22.22.22.22.
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SIP SETTINGS
The UCM6100 SIP global settings can be accessed via Web GUI->PBX->SIP Settings.
SIP SETTINGS/GENERAL
Table 55: 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
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
Allow Transfer
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.
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From Domain
Auto Domain
Allow External Domains
Configure the domain in the "From:" header of the SIP message. It may
be required by some providers for authentication.
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 56: 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
Register Attempts
gives 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
Reject Non-Matching INVITE
notifying the requester whether there is a matching user or peer for the
request. This reduces the ability of an attacker to scan for valid SIP
usernames. The default setting is "No".
SIP SETTINGS/SESSION TIMER
Table 57: 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.
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
Refuse
Do not run session timer.
Session Expire
Min SE
Session Refresher
Configure the maximum session refresh interval (in seconds). The
default setting is 1800.
Configure the minimum session refresh interval (in seconds). The
default setting is 90.
Select the session refresher to be UAC or UAS. The default setting is
UAC.
SIP SETTINGS/TCP and TLS
Table 58: 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
TCP Bind Address
binding to all interfaces. The port number is optional. If not specified,
5060 will be used.
TLS Enable
Configure to allow incoming TLS connections with the 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
TLS Self-Signed CA
self-signed certificate, including server's public key. This file will be
renames as "TLS.ca" automatically.
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Note:
The size of the uploaded ca file must be under 2MB.
This is the Certificate file (*.pem format only) used for TLS connections.
It contains private key for client and signed certificate for the server. This
TLS Cert
file will be renamed as "TLS.pem" automatically.
Note:
The size of the uploaded certificate file must be under 2MB.
This file must be named with the CA subject name hash value. It
contains CA's (Certificate Authority) public key, which is used to verify
TLS CA Cert
the accessed servers.
Note:
The size of the uploaded CA certificate file must be under 2MB.
TLS CA List
Display a list of files under the CA Cert directory.
SIP SETTINGS/NAT
Table 59: 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:
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192.168.0.0/16
SIP SETTINGS/TOS
Table 60: 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
Time
and 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
DTMF Mode
selected, 64-kbit codec PCMU and PCMA are required. When "Auto" is
selected, "RFC2833" will be used if offered, otherwise "Inband" will be
used. The default setting is "RFC2833".
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
Firmware Version 1.0.6.10
Configure whether the Remote-Party-ID should be trusted. The default
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setting is "No".
Send Remote Party ID
Configure whether the Remote-Party-ID should be sent or not. The
default setting is "No".
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 1.0.6.10
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 69: Status->PBX Status
TRUNKS
Users could see all the configured trunk status in this section.
Figure 70: Trunk Status
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Table 61: Trunk Status
Display trunk status.

Analog trunk status:
Available
Busy
Unavailable
Unknown Error

Status
SIP Peer trunk status:
Unreachable: The hostname cannot be reached.
Unmonitored: 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.
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Figure 71: Extension Status
Table 62: Extension Status
Display extension number (including feature code). The color indicator has
the following definitions.
Status

Green: Free

Blue: Ringing

Yellow: In Use

Grey: Unavailable
Extension
Display the extension number.
Name/Label
Display name (callerID name) or label for the extension.
Display message status for the extension.
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

Features
Other operations are also available in extension status section:

Click on "Extensions", the web page will redirect to extension configuration page which can also be
accessed via web GUI->PBX->Basic/Call Routes->Extensions.

Click on
to refresh the extension status.
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
Click on one of the tabs
to display the corresponding extensions
accordingly.

Click on [ + ] to expand the status detail table.

Click on [ - ] to hide the status detail table.
QUEUES
Users could see all the configured call queue status in this section. The following figure shows the call
queue 6500 being in used.
Figure 72: 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 63: 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.
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CONFERENCE ROOMS
Users could see all the conference room status in this section. It shows all the configured conference
rooms, current users, call duration for each user and conference call.
Figure 73: 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 hide the conference room details.
to refresh the conference room status.
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 74: UCM6116 Interfaces Status
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Table 64: Interface Status Indicators
USB connected.
USB disconnected.
SD Card connected.
SD Card disconnected.
LAN/WAN connected.
LAN/WAN not configured.
LAN/WAN disconnected.
FXS/FXO connected.
FXS/FXO waiting.
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 75: Parking Lot Status
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Table 65: 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.
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.
ACTIVITY CALLS
The UCM6100 can monitor the status of active calls in real time. The active calls status can be viewed
under web GUI->Status->Active Calls.
The following figure shows 1001 Jane Doe is calling 1002 William Tsai. 1002 is ringing.
Figure 76: Status->PBX Status->Activity Calls: Calling
The following figure shows the call between 1001 Jane Doe and 1002 William Tsai is established.
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Figure 77: Status->PBX Status->Activity Calls

Click on

Click on

Click on
to refresh the active call status.
to hang up all calls.
to hang up single call.
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 66: 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.
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Up Time
System up time since the last reboot.
Idle Time
System idle time since the last reboot.
Boot
Boot version.
Core
Core version.
Base
Base version.
Program
Program version. This is the main software release version.
Recovery
Recovery version.
NETWORK
Under Web GUI->Status->System Status->Network, users could check the network information for the
UCM6100. Please see details in the following table.
Table 67: 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 78: 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 79: 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:
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
Disk Usage
Modify Admin Password
Memory Usage
System Reboot
System Update
System Crash
Click on
to configure the parameters for each event. See examples below.
1. Disk Usage.
Figure 80: 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 81: 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 82: 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 83: 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 when the alert is
triggered for the configured system events. The following picture shows disk usage alert log. We can tell
the detect cycle for the disk usage is 10 minutes and the disk usage is restored to normal after the
administrator cleans up the disk storage below the threshold.
Figure 84: System Events->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.
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
Talk Time. Format: 0:00:10

Status. Format: NO ANSWER, BUSY, ANSWERED, or FAILED.

Options. Voice record playing/downloading/deleting.
Users could filter the call report by specifying the date range and criteria, depending on how the users
would like to include the logs to the report. Then click on "View Report" button to display the generated
report.
Figure 85: CDR Filter
Table 68: 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.
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The call report will display as the following figure shows.
Figure 86: Call Report
Users could perform the following operations on the call report.

Sort
Click on the header of the column to sort by this category. For example, clicking on "Start Time" will
sort the report according to start time. Clicking on "Start Time" again will reverse the order.

Download Records
On the bottom of the page, click on "Download Records" button to export the report in .csv format.

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 87: Call Report Entry With Audio Recording File
DOWNLOADED CDR FILE
The downloaded CDR (.csv file) has different format from the web UI CDR. Here are some descriptions.

Call From, Call To
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"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 88: 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:
Figure 89: 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:
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Figure 90: 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 91: 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
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.
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Figure 92: CDR Statistics
Table 69: 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 93: CDR->Recording Files

Click on
to play the recording file.

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.
CDR API CONFIGURATION FILES
The UCM6100 supports third party billing interface API for external billing software to access CDR on the
PBX. The API uses HTTPS to request the CDR 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.
Table 70: CDR API Configuration Files
Enable
Enable/Disable CDR API. The default setting is disabled.
TLS Bind Address
Configure the IP address for TLS server to bind to. "0.0.0.0" means binding to all
interfaces. The port number is optional and the default port number is 8443. The IP
address must match the common name (host name) in the certificate so that the
TLS socket won't bind to multiple IP addresses. The default setting is 0.0.0.0:8443.
TLS Private Key
Upload TLS private key. The size of the key file must be under 2MB. This file will be
renamed as 'private.pem' automatically.
TLS Cert
Upload TLS cert. The size of the certificate must be under 2MB. This is the
certificate file (*.pem format only) for TLS connection. This file will be renamed as
"certificate.pem" automatically. It contains private key for the client and signed
certificate for the server.
TLS Authentication
Configure the user name for TLS authentication. If not configured, authentication
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Name
will be skipped.
TLS Authentication
Password
Permitted
Configure the password for TLS authentication. This is optional.
Specify a list of IP addresses permitted by CDR 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 format of the HTTPS request for the CDR API is as below.
https://[UCM IP]:[Port]/cdrapi?[option1]=[value]&[option2]=[value]&...
By default, the port number for the API is 8443.
The options included in the request URI control the record matching and output format. For CDR matching
parameters, all non-empty parameters must have a match to return a record. Parameters can appear in
the URI in any order. Multiple values given for caller or callee will be concatenated. The following table
shows the parameter list used in the CDR API.
Table 71: CDR API URI Parameters
Field
Value
format
csv, xml, json
Details
Define the format for output of matching CDR rows.
Default is csv (comma separated values).
Number of records to return. Default is 1000, which is
numRecords
Number: 0-1000
also the maximum allowed value.
Number of matching records to skip. This will be
offset
combined with numRecords to receive all matches
Number
over multiple responses. Default is 0.
caller
Comma separated extensions,
Filters based on src (caller) or dst (callee) value,
ranges of extensions, or regular
matching any extension contained in the parameter
expressions.
input string.
Example:
Patterns containing one or more wildcards ('@' or '_')
will match as a regular expression, and treat '-' as a
callee
caller=5300,5302-5304,[email protected]
-OR-
literal hyphen rather than a range signifier. The '@'
wildcard matches any number of characters (including
zero), while '_' matches any single character.
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caller=5300&caller=5302-5304&
Otherwise, patterns containing a single hyphen will be
[email protected]
matching a range of numerical extensions, with
(Matches extensions 5300, 5302,
5303, 5304, and any extension
non-numerical characters ignored, while patterns
containing multiple hyphens will be ignored. (The
containing 4 as the second
pattern "0-0" will match all non-numerical and empty
digit/character).
strings).
Date and/or time of day in any of
startTime
the following formats:
YYYY-MM-DDTHH:MM
Filters based on the start (call start time) value. Calls
which start within this period (inclusive of boundaries)
YYYY-MM-DDTHH:MM:SS
YYYY-MM-DDTHH:MM:SS.SSS
will match, regardless of the call answer or end time.
An empty value for either field will be interpreted as
range with no minimum or maximum respectively.
(literal 'T' character separator in
Strings without a date have a default value of
above three formats)
endTime
2000-01-01. Strings without a time of day have a
HH:MM
default value of of 00:00 UTC, while strings with a time
of day specified may also optionally specify a time
HH:MM:SS
zone offset - replace '+' in time zone offset with '%2B'
HH:MM:SS.SSS
(see http://www.w3.org/TR/NOTE-datetime).
now
DDDDDDDDDD
minDur
Filters based on the billsec value, the duration between
Number (duration in seconds)
call answer and call end.
maxDur
Example Queries:
The following illustrates the format of queries to accomplish certain requests. In most cases, multiple
different queries will accomplish the same goal, and these examples are not intended to be exhaustive, but
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rather to bring attention to particular features of the CDR API connector.
Query 1: Request all records of calls placed on extension 5300 which last between 8 and 60 seconds
(inclusive), with results in CSV format.
https://192.168.254.200:8088/cdrapi?format=CSV&caller=5300&minDur=8&maxDur=60
-ORhttps://192.168.254.200:8088/cdrapi?caller=5300&minDur=8&maxDur=60
Query 2: Request all records of calls placed on extension 5300 or in the range 6300-6399 to extensions
starting with 5, with results in XML format.
https://192.168.254.200:8088/cdrapi?format=XML&caller=5300,6300-6399&[email protected]
-ORhttps://192.168.254.200:8088/cdrapi?cdrapi?format=XML&caller=5300&caller=6300-6399&[email protected]
Query 3: Request all records of calls placed on extensions containing substring "53" prior to January 23,
2013 00:00:00 UTC to extensions 5300-5309, with results in CSV format.
https://192.168.254.200:8088/[email protected]@&callee=5300-5309&endTime=2013-01-23
-ORhttps://192.168.254.200:8088/[email protected]@&callee=530_&endTime=2013-01-23T00:00:00
Query 4: Request all records of calls placed by an Anonymous caller during July 2013 Central Standard
Time to extensions starting with 2 or 34 or ending with 5, with results in CSV format.
https://192.168.254.200:8088/cdrapi?caller=Anonymous&[email protected],[email protected],@5&startTime=2013-07-01T00:00:0006:00&endTime=2013-07-31T23:59:59-06:00
Query 5: Request all records during July 2013 Central Standard Time, 200 at a time, with results in CSV
format.
https://192.168.254.200:8088/cdrapi?startTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:59-06:
00&numRecords=200&offset=0
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-THENhttps://192.168.254.200:8088/cdrapi?sstartTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:59-0
6:00&numRecords=200&offset=200
-THENhttps://192.168.254.200:8088/cdrapi?startTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:59-06:
00&numRecords=200&offset=400
Note:

Disallowed characters in the caller, callee, startTime, or endTime strings, and non-digit characters
in the values of numRecords, offset, minDur, or maxDur, will result in no records returned - the
appropriate container/header for the output format will be the only output. If the format parameter
is in error, the CSV header will be used. Error messages will appear in the Asterisk log (along with
errors stemming from failed database connections, etc.).

Other errors which return no records include:
- Multiple hyphens in an extension range (e.g. caller=5300-5301-,6300)
- Empty parameter value (e.g. caller=)
- Extension values starting with comma, or with consecutive commas (e.g. caller=5300,,5303)
- Unknown parameters (e.g. caler=5300) or URI ending with '&'
- Except for caller and callee, multiple instances of the same parameter within the URI (e.g.
minDur=5&minDur=10)
Example Output:
The following are examples of each of the output formats for the same data set.
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CSV:
AcctId,accountcode,src,dst,dcontext,clid,channel,dstchannel,lastapp,lastdata,start,answer,end,duration
,billsec,disposition,amaflags,uniqueid,userfield,channel_ext,dstchannel_ext,service
62,,5300,5301,from-internal,"pn01"
<5300>,SIP/5300-00000000,SIP/5301-00000001,Dial,SIP/5301,60,,2013-12-03 11:46:40,2013-12-03
11:46:43,2013-12-03 11:46:49,9,6,ANSWERED,DOCUMENTATION,1386092800.0,EXT,5300,5301,s
63,,5300,5301,from-internal,"pn01"
<5300>,SIP/5300-00000000,SIP/5301-00000001,Dial,SIP/5301,60,,2013-12-03 14:01:41,2013-12-03
14:01:43,2013-12-03 14:01:46,5,3,ANSWERED,DOCUMENTATION,1386100901.0,EXT,5300,5301,s
64,,5300,5301,from-internal,"pn01"
<5300>,SIP/5300-00000002,SIP/5301-00000003,Dial,SIP/5301,60,,2013-12-03 14:02:23,2013-12-03
14:02:27,2013-12-03 14:02:31,8,4,ANSWERED,DOCUMENTATION,1386100943.2,EXT,5300,5301,s
XML:
<root>
<cdr><AcctId>62</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext
>from-internal</dcontext><clid>&quot;pn01&quot;
&lt;5300&gt;</clid><channel>SIP/5300-00000000</channel><dstchannel>SIP/5301-00000001</dstcha
nnel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03
11:46:40</start><answer>2013-12-03 11:46:43</answer><end>2013-12-03
11:46:49</end><duration>9</duration><billsec>6</billsec><disposition>ANSWERED</disposition><a
maflags>DOCUMENTATION</amaflags><uniqueid>1386092800.0</uniqueid><userfield>EXT</userfi
eld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service>
</cdr>
<cdr><AcctId>63</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext
>from-internal</dcontext><clid>&quot;pn01&quot;
&lt;5300&gt;</clid><channel>SIP/5300-00000000</channel><dstchannel>SIP/5301-00000001</dstcha
nnel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03
14:01:41</start><answer>2013-12-03 14:01:43</answer><end>2013-12-03
14:01:46</end><duration>5</duration><billsec>3</billsec><disposition>ANSWERED</disposition><a
maflags>DOCUMENTATION</amaflags><uniqueid>1386100901.0</uniqueid><userfield>EXT</userfi
eld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service>
</cdr>
<cdr><AcctId>64</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext
>from-internal</dcontext><clid>&quot;pn01&quot;
&lt;5300&gt;</clid><channel>SIP/5300-00000002</channel><dstchannel>SIP/5301-00000003</dstcha
nnel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03
14:02:23</start><answer>2013-12-03 14:02:27</answer><end>2013-12-03
14:02:31</end><duration>8</duration><billsec>4</billsec><disposition>ANSWERED</disposition><a
maflags>DOCUMENTATION</amaflags><uniqueid>1386100943.2</uniqueid><userfield>EXT</userfi
eld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service>
</cdr>
</root>
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JSON:
{
"cdr":
[
{ "AcctId": "62", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal", "clid":
"\"pn01\" <5300>", "channel": "SIP/5300-00000000", "dstchannel": "SIP/5301-00000001", "lastapp":
"Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 11:46:40", "answer": "2013-12-03 11:46:43",
"end": "2013-12-03 11:46:49", "duration": "9", "billsec": "6", "disposition": "ANSWERED", "amaflags":
"DOCUMENTATION", "uniqueid": "1386092800.0", "userfield": "EXT", "channel_ext": "5300",
"dstchannel_ext": "5301", "service": "s" },
{ "AcctId": "63", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal", "clid":
"\"pn01\" <5300>", "channel": "SIP/5300-00000000", "dstchannel": "SIP/5301-00000001", "lastapp":
"Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 14:01:41", "answer": "2013-12-03 14:01:43",
"end": "2013-12-03 14:01:46", "duration": "5", "billsec": "3", "disposition": "ANSWERED", "amaflags":
"DOCUMENTATION", "uniqueid": "1386100901.0", "userfield": "EXT", "channel_ext": "5300",
"dstchannel_ext": "5301", "service": "s" },
{ "AcctId": "64", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal", "clid":
"\"pn01\" <5300>", "channel": "SIP/5300-00000002", "dstchannel": "SIP/5301-00000003", "lastapp":
"Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 14:02:23", "answer": "2013-12-03 14:02:27",
"end": "2013-12-03 14:02:31", "duration": "8", "billsec": "4", "disposition": "ANSWERED", "amaflags":
"DOCUMENTATION", "uniqueid": "1386100943.2", "userfield": "EXT", "channel_ext": "5300",
"dstchannel_ext": "5301", "service": "s" }
]
}
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UPGRADING AND MAINTENANCE
UPGRADING
The 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 94: Network Upgrade
Table 72: Network Upgrade Configuration
Upgrade Via
Allow users to choose the firmware upgrade method: TFTP, HTTP or
HTTPS.
Firmware Server Path
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Define the server path for the firmware server.
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Firmware File Prefix
If configured, only the firmware with the matching encrypted prefix will
be downloaded and flashed into the UCM6100.
Firmware File Suffix
If configured, only the firmware with the matching encrypted postfix will
be 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 95: Local Upgrade

Click on
to start upgrading.
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Figure 96: Upgrading Firmware Files

Wait until the upgrading process is successful and a window will be popped up in the Web GUI.
Figure 97: 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 98: 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 99: Data Sync
Table 73: 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.
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.
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Figure 100: 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.
CLEANER
Users could configure to clean the Call Detail Report/Voice Records/Voice Mails/FAX automatically under
Web GUI->Maintenance->Cleaner.
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Figure 101: Cleaner
Table 74: 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.
VR Clean Threshold
Specify the Voice Records threshold from 0 to 99 by using local storage
status in percentage.
VR Clean Time
Enter 0-23 to specify the hour of the day to clean up Voice Records.
Clean Interval
Enter 1-30 to specify the day of the month to clean up Voice Records.
All the cleaner logs will be listed on the bottom of the page.
RESET AND REBOOT
Users could perform reset and reboot under Web GUI->Maintenance->Reset and Reboot.
To factory reset the device, select the mode type first. There are two different types for reset.

User Data: All the data including voicemail, recordings, IVR Prompt, Music on Hold, CDR and backup
files will be cleared.

All: All the configurations and data will be reset to factory default.
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Figure 102: 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.
TROUBLESHOOTING
On the UCM6100, users could capture traces, ping remote host and traceroute remote host for
troubleshooting purpose under Web GUI->Maintenance->Troubleshooting.
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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 103: Ethernet Capture
The output result is in .pcap format. Therefore, users could specify the capture filter as used in general
network traffic capture tool (host, src, dst, net, protocol, port, port range) before starting capturing the
trace.
IP PING
Enter the target host in host name or IP address. Then press "Start" button. The output result will
dynamically display in the window below.
Figure 104: PING
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TRACEROUTE
Enter the target host in host name or IP address. Then press "Start" button. The output result will
dynamically display in the window below.
Figure 105: Traceroute
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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).
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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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