Grandstream Networks UCM6104 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 1.0.5.19 ........................................................................................................ 12


FIRMWARE VERSION 1.0.4.7 .......................................................................................................... 12




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 ................................................................................... 25

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 ................................................................................................................ 30

SYSTEM SETTINGS ................................................................................... 32

NETWORK SETTINGS ...................................................................................................................... 32

Firmware Version 1.0.5.19

UCM6100 Series IP PBX User Manual Page 1 of 183

BASIC SETTINGS ...................................................................................................................... 32

802.1X ........................................................................................................................................ 36

PORT FORWORDING (UCM6102 ONLY) .................................................................................. 37

FIREWALL ......................................................................................................................................... 37

STATIC DEFENSE...................................................................................................................... 37

DYNAMIC DEFENSE ................................................................................................................. 40

FAIL2BAN ................................................................................................................................... 40

CHANGE PASSWORD ...................................................................................................................... 41

LDAP SERVER .................................................................................................................................. 42

LDAP SERVER CONFIGURATIONS .......................................................................................... 42

LDAP PHONEBOOK .................................................................................................................. 43

LDAP CLIENT CONFIGURATIONS ............................................................................................ 45

HTTP SERVER .................................................................................................................................. 46

EMAIL SETTINGS ............................................................................................................................. 47

TIME SETTINGS ............................................................................................................................... 48

NTP SERVER .................................................................................................................................... 50

PROVISIONING ........................................................................................... 51

OVERVIEW ....................................................................................................................................... 51

AUTO PROVISIONING ...................................................................................................................... 51

MANUAL PROVISIONING ................................................................................................................. 54

DISCOVERY ............................................................................................................................... 54

ASSIGNMENT ............................................................................................................................ 55

CREATE NEW DEVICE .............................................................................................................. 55

PROVISIONING ......................................................................................................................... 56

EXTENSIONS .............................................................................................. 57

CREATE NEW USER ........................................................................................................................ 57

CREATE NEW SIP EXTENSION ................................................................................................ 57

CREATE NEW IAX EXTENSION ................................................................................................ 60

CREATE NEW FXS EXTENSION .............................................................................................. 63

BATCH ADD EXTENSIONS .............................................................................................................. 66

BATCH ADD SIP EXTENSIONS ................................................................................................. 66

BATCH ADD IAX EXTENSIONS................................................................................................. 69

EDIT EXTENSION ............................................................................................................................. 71

TRUNKS ...................................................................................................... 73

ANALOG TRUNKS ............................................................................................................................ 73

ANALOG TRUNK CONFIGURATION ......................................................................................... 73

PSTN DETECTION .................................................................................................................... 75

VOIP TRUNKS ................................................................................................................................... 78



Direct Outward Dialing (DOD)..................................................................................................... 86

CALL ROUTES ........................................................................................... 88

OUTBOUND ROUTES ...................................................................................................................... 88

INBOUND ROUTES .......................................................................................................................... 90

INBOUND RULE CONFIGURATIONS ....................................................................................... 90

BLACKLIST CONFIGURATIONS ............................................................................................... 93

CONFERENCE BRIDGE ............................................................................ 94

CONFERENCE BRIDGE CONFIGURATIONS ........................................................................... 94

JOIN A CONFERENCE CALL .................................................................................................... 96

INVITE OTHER PARTIES TO JOIN CONFERENCE .................................................................. 96

DURING THE CONFERENCE ................................................................................................... 97

RECORD CONFERENCE .......................................................................................................... 98

IVR ............................................................................................................. 100

CONFIGURE IVR ............................................................................................................................ 100

CREATE IVR PROMPT ................................................................................................................... 101

RECORD NEW IVR PROMPT.................................................................................................. 102

UPLOAD IVR PROMPT ............................................................................................................ 103

LANGUAGE SETTINGS FOR VOICE PROMPT ..................................... 104

DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE .............................................................. 104

CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE ............................................................. 107

VOICEMAIL ............................................................................................... 108

CONFIGURE VOICEMAIL ............................................................................................................... 108

VOICEMAIL EMAIL SETTINGS ....................................................................................................... 109

CONFIGURE VOICEMAIL GROUP ..................................................................................................110

RING GROUP ............................................................................................. 112

CONFIGURE RING GROUP ............................................................................................................112

PAGING AND INTERCOM GROUP .......................................................... 114

CONFIGURE PAGING/INTERCOM GROUP....................................................................................114

CALL QUEUE ............................................................................................. 116

CONFIGURE CALL QUEUE .............................................................................................................116

EXTENSION GROUPS .............................................................................. 119

CONFIGURE EXTENSION GROUPS ..............................................................................................119



USE EXTENSION GROUPS ............................................................................................................119

PICKUP GROUPS ..................................................................................... 121

CONFIGURE PICKUP GROUPS..................................................................................................... 121

MUSIC ON HOLD ...................................................................................... 122

FAX/T.38 .................................................................................................... 123

CONFIGURE FAX/T.38 .................................................................................................................... 123

SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE .............................................. 124

SAMPLE CONFIGURATION FOR FAX-TO-EMAIL ......................................................................... 126

DISA ........................................................................................................... 128

BLF AND EVENT LIST ............................................................................. 130

BLF .................................................................................................................................................. 130

EVENT LIST .................................................................................................................................... 130

CALL FEATURES ..................................................................................... 133

FEATURE CODES ........................................................................................................................... 133

CALL RECORDING ......................................................................................................................... 136

CALL PARK ..................................................................................................................................... 137

PARK A CALL ........................................................................................................................... 137

RETRIEVE THE PARKED CALL .............................................................................................. 137

INTERNAL OPTIONS ............................................................................... 138

INTERNAL OPTIONS/GENERAL .................................................................................................... 138

INTERNAL OPTIONS/JITTER BUFFER .......................................................................................... 139

INTERNAL OPTIONS/RTP SETTINGS ........................................................................................... 140

INTERNAL OPTIONS/HARDWARE CONFIG.................................................................................. 140

INTERNAL OPTIONS/STUN MONITOR .......................................................................................... 142

IAX SETTINGS .......................................................................................... 143

IAX SETTINGS/GENERAL .............................................................................................................. 143

IAX SETTINGS/REGISTRATION..................................................................................................... 143

IAX SETTINGS/STATIC DEFENSE ................................................................................................. 144

SIP SETTINGS .......................................................................................... 146

SIP SETTINGS/GENERAL .............................................................................................................. 146

SIP SETTINGS/MISC ...................................................................................................................... 147

SIP SETTINGS/SESSION TIMER ................................................................................................... 147



SIP SETTINGS/TCP and TLS .......................................................................................................... 148

SIP SETTINGS/NAT ........................................................................................................................ 149

SIP SETTINGS/TOS ........................................................................................................................ 150

STATUS AND REPORTING ...................................................................... 153

PBX STATUS ................................................................................................................................... 153

TRUNKS ................................................................................................................................... 153

EXTENSIONS........................................................................................................................... 154

QUEUES .................................................................................................................................. 156

CONFERENCE ROOMS .......................................................................................................... 157

INTERFACES STATUS ............................................................................................................. 157

PARKING LOT .......................................................................................................................... 158

SYSTEM STATUS ........................................................................................................................... 159

GENERAL ................................................................................................................................ 159

NETWORK ............................................................................................................................... 160

STORAGE USAGE ................................................................................................................... 160

RESOURCE USAGE ................................................................................................................ 161

SYSTEM EVENTS ........................................................................................................................... 162

ALERT EVENTS LIST .............................................................................................................. 162

ALERT LOG .............................................................................................................................. 164

ALERT CONTACT .................................................................................................................... 164

CDR ................................................................................................................................................. 165

DOWNLOADED CDR FILE ...................................................................................................... 167

STATISTICS ............................................................................................................................. 169

RECORDING FILES ................................................................................................................. 170

UPGRADING AND MAINTENANCE ........................................................ 171

UPGRADING ................................................................................................................................... 171

UPGRADING VIA NETWORK .................................................................................................. 171

UPGRADING VIA LOCAL UPLOAD ......................................................................................... 172

NO LOCAL FIRMWARE SERVERS ......................................................................................... 174

BACKUP .......................................................................................................................................... 174

LOCAL BACKUP ...................................................................................................................... 174

DATA SYNC .............................................................................................................................. 175

RESTORE CONFIGURATION FROM BACKUP FILE .............................................................. 176

CLEANER ........................................................................................................................................ 177

RESET AND REBOOT .................................................................................................................... 178

SYSLOG .......................................................................................................................................... 178

TROUBLESHOOTING ..................................................................................................................... 179

ETHERNET CAPTURE ............................................................................................................ 179

IP PING..................................................................................................................................... 179



TRACEROUTE ......................................................................................................................... 180

EXPERIENCING THE UCM6100 SERIES IP PBX .................................. 181



Table of Tables


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 Firewall->Static Defense->Current Service............................................................... 38

Table 13: Typical Firewall Settings ............................................................................................................ 38

Table 14: Firewall Rule Settings ................................................................................................................ 39

Table 15: UCM6102 Firewall Dynamic Defense ........................................................................................ 40

Table 16: Fail2Ban Settings ...................................................................................................................... 41

Table 17: HTTP Server Settings ................................................................................................................ 47

Table 18: Email Settings............................................................................................................................ 47

Table 19: Time Auto Updating ................................................................................................................... 48

Table 20: Auto Provision Settings .............................................................................................................. 53

Table 21: SIP Extension Configuration Parameters .................................................................................. 57

Table 22: IAX Extension Configuration Parameters .................................................................................. 60

Table 23: FXS Extension Configuration Parameters ................................................................................. 63

Table 24: Batch Add SIP Extension Parameters ....................................................................................... 66

Table 25: Batch Add IAX Extension Parameters ....................................................................................... 69

Table 26: Analog Trunk Configuration Parameters .................................................................................... 73

Table 27: PSTN Detection For Analog Trunk............................................................................................. 78

Table 28: SIP Trunk Configuration Parameters ......................................................................................... 79

Table 29: IAX Trunk Configuration Parameters ......................................................................................... 83

Table 30: Outbound Route Configuration Parameters .............................................................................. 88

Table 31: Inbound Rule Configuration Parameters ................................................................................... 90

Table 32: Conference Bridge Configuration Parameters ........................................................................... 94

Table 33: Conference Caller IVR Menu ..................................................................................................... 98

Table 34: IVR Configuration Parameters ................................................................................................. 100

Table 35: Voicemail Settings ................................................................................................................... 108

Table 36: Voicemail Email Settings ..........................................................................................................110

Table 37: Voicemail Group Settings ......................................................................................................... 111

Table 38: Ring Group Parameters ............................................................................................................112



Table 39: Paging/Intercom Group Configuration Parameters ...................................................................114

Table 40: Call Queue Configuration Parameters ......................................................................................116

Table 41: FAX/T.38 Settings .................................................................................................................... 123

Table 42: DISA Settings .......................................................................................................................... 128

Table 43: Event List Settings ................................................................................................................... 131

Table 44: UCM6100 Feature Codes ........................................................................................................ 133

Table 45: Internal Options/General ......................................................................................................... 138

Table 46: Internal Options/Jitter Buffer .................................................................................................... 140

Table 47: Internal Options/RTP Settings ................................................................................................. 140

Table 48: Internal Options/Hardware Config ........................................................................................... 141

Table 49: Internal Options/STUN Monitor................................................................................................ 142

Table 50: IAX Settings/General ............................................................................................................... 143

Table 51: IAX Settings/Registration ......................................................................................................... 143

Table 52: IAX Settings/Static Defense ..................................................................................................... 144

Table 53: SIP Settings/General ............................................................................................................... 146

Table 54: SIP Settings/Misc .................................................................................................................... 147

Table 55: SIP Settings/Session Timer ..................................................................................................... 147

Table 56: SIP Settings/TCP and TLS ...................................................................................................... 148

Table 57: SIP Settings/NAT ..................................................................................................................... 149

Table 58: SIP Settings/ToS ...................................................................................................................... 150

Table 59: Trunk Status............................................................................................................................. 154

Table 60: Extension Status ...................................................................................................................... 155

Table 61: Agent Status ............................................................................................................................ 156

Table 62: Interface Status Indicators ....................................................................................................... 158

Table 63: Parking Lot Status ................................................................................................................... 159

Table 64: System Status->General.......................................................................................................... 159

Table 65: System Status->Network ......................................................................................................... 160

Table 66: CDR Filter Criteria ................................................................................................................... 165

Table 67: CDR Statistics Filter Criteria .................................................................................................... 169

Table 68: Network Upgrade Configuration .............................................................................................. 171

Table 69: Data Sync Configuration .......................................................................................................... 176

Table 70: Cleaner Configuration .............................................................................................................. 177



Table of Figures


Figure 1: UCM6102 Front View ................................................................................................................. 21

Figure 2: UCM6102 Back View ................................................................................................................. 21







Figure 9: UCM6116 Web GUI Login Page ................................................................................................ 28

Figure 10: UCM6100 Web GUI Language ................................................................................................ 30

Figure 11: Create New Firewall Rule......................................................................................................... 39

Figure 12: LDAP Server Configurations .................................................................................................... 42

Figure 13: Default LDAP Phonebook DN .................................................................................................. 43

Figure 14: Default LDAP Phonebook Attributes ........................................................................................ 43

Figure 15: Add LDAP Phonebook ............................................................................................................. 44

Figure 16: Edit LDAP Phonebook ............................................................................................................. 44

Figure 17: GXP2200 LDAP Phonebook Configuration .............................................................................. 46

Figure 18: UCM6100 Email Settings ......................................................................................................... 48

Figure 19: Set Time Manually ................................................................................................................... 50

Figure 20: UCM6100 Zero Config ............................................................................................................. 52

Figure 21: Auto Provision Settings ............................................................................................................ 53

Figure 22: Auto Discover ........................................................................................................................... 54

Figure 23: Discovered Devices ................................................................................................................. 55

Figure 24: Assign Extension To Device ..................................................................................................... 55

Figure 25: Create New Device .................................................................................................................. 56

Figure 26: UCM6100 FXO Tone Settings .................................................................................................. 76

Figure 27: UCM6100 PSTN Detection ...................................................................................................... 76

Figure 28: UCM6100 PSTN Detection: Auto Detect .................................................................................. 77

Figure 29: UCM6100 PSTN Detection: Semi-Auto Detect ........................................................................ 77

Figure 30: DOD extension selection ......................................................................................................... 86

Figure 31: Edit DOD .................................................................................................................................. 87

Figure 32: Blacklist Configuration Parameters .......................................................................................... 93

Figure 33: Conference Invitation From Web GUI ...................................................................................... 96

Figure 34: Conference Recording ............................................................................................................. 99

Figure 35: Click On Prompt To Create IVR Prompt ................................................................................. 102

Figure 36: Record New IVR Prompt ........................................................................................................ 102

Figure 37: Upload IVR Prompt ................................................................................................................ 103

Figure 38: Language Settings For Voice Prompt .................................................................................... 105



Figure 39: Voice Prompt Package List .................................................................................................... 106

Figure 40: New Voice Prompt Language Added ..................................................................................... 107

Figure 41: Voicemail Email Settings ........................................................................................................ 109

Figure 42: Voicemail Group......................................................................................................................110

Figure 43: Ring Group .............................................................................................................................112

Figure 44: Ring Group Configuration .......................................................................................................113

Figure 45: Paging/Intercom Group ...........................................................................................................114

Figure 46: Page/Intercom Group Settings ................................................................................................115

Figure 47: Call Queue ..............................................................................................................................116

Figure 48: Agent Login Settings ...............................................................................................................118

Figure 49: Edit Extension Group ..............................................................................................................119

Figure 50: Select Extension Group in Outbound Route .......................................................................... 120

Figure 51: Edit Pickup Group .................................................................................................................. 121

Figure 52: Music On Hold Default Class ................................................................................................. 122

Figure 53: Configure Analog Trunk without Fax Detection ...................................................................... 124

Figure 54: Configure Extension For Fax Machine ................................................................................... 125

Figure 55: Configure Inbound Rule For Fax ............................................................................................ 125

Figure 56: Create Fax Extension ............................................................................................................ 126

Figure 57: Enable Fax Detection In Analog Trunk ................................................................................... 126

Figure 58: Inbound Route To Fax Extension ........................................................................................... 127

Figure 59: Create New DISA................................................................................................................... 128

Figure 60: Create New Event List ........................................................................................................... 131

Figure 61: Download Recording File From CDR Page ........................................................................... 136

Figure 62: FXS Ports Signaling Preference ............................................................................................ 141

Figure 63: FXO Ports ACIM Settings....................................................................................................... 141

Figure 64: Status->PBX Status................................................................................................................ 153

Figure 65: Trunk Status ........................................................................................................................... 153

Figure 66: Extension Status .................................................................................................................... 155

Figure 67: Queue Status ......................................................................................................................... 156

Figure 68: Conference Room Status ....................................................................................................... 157

Figure 69: UCM6116 Interfaces Status ................................................................................................... 157

Figure 70: Parking Lot Status .................................................................................................................. 158

Figure 71: System Status->Storage Usage ............................................................................................. 161

Figure 72: System Status->Resource Usage .......................................................................................... 161

Figure 73: System Events->Alert Events Lists: Disk Usage .................................................................... 162

Figure 74: System Events->Alert Events Lists: Modify Admin Password ................................................ 162

Figure 75: System Events->Alert Events Lists: Memory Usage .............................................................. 163

Figure 76: System Events->Alert Events Lists: System Reboot .............................................................. 163

Figure 77: System Events->Alert Events Lists: System Update .............................................................. 163

Figure 78: System Events->Alert Events Lists: System Crash ................................................................ 164

Figure 79: System Events->Alert Log ..................................................................................................... 164



Figure 80: CDR Filter .............................................................................................................................. 165

Figure 81: Call Report ............................................................................................................................. 166

Figure 82: Call Report Entry With Audio Recording File ......................................................................... 167

Figure 83: Downloaded CDR File Sample - Call To Shows "s" ............................................................... 167

Figure 84: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ................................. 168



Figure 87: CDR Statistics ........................................................................................................................ 169

Figure 88: Network Upgrade ................................................................................................................... 171

Figure 89: Local Upgrade........................................................................................................................ 172

Figure 90: Upgrading Firmware Files ...................................................................................................... 173

Figure 91: Reboot UCM6100 .................................................................................................................. 173

Figure 92: Local Backup ......................................................................................................................... 175

Figure 93: Data Sync .............................................................................................................................. 175

Figure 94: Restore UCM6100 From Backup File .................................................................................... 176

Figure 95: Cleaner .................................................................................................................................. 177

Figure 96: Reset and Reboot .................................................................................................................. 178

Figure 97: Ethernet Capture.................................................................................................................... 179

Figure 98: PING ...................................................................................................................................... 180

Figure 99: Traceroute .............................................................................................................................. 180



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 1.0.5.19



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 44: UCM6100 Feature Codes]



Changed valid range for option "Current Disconnect Threshold". [Table 26: Analog Trunk

Configuration Parameters]




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 30: 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 38: 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 45: Internal Options/General]



Added sample descriptions for downloaded CDR file. [DOWNLOADED CDR FILE]




Asterisk updated to version 1.8.23.1.



Added DID routing support for incoming calls. [Table 28: SIP Trunk Configuration Parameters]



Added DOD routing support. [Direct Outward Dialing (DOD)]





Added GXP one-button Voicemail access. [Table 21: SIP Extension Configuration Parameters]



Added option "Skip voicemail password verification" on extension edit page. [Table 21: SIP

Extension Configuration Parameters]



Added Hot-Desking Support. [Table 21: 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 48:

Internal Options/Hardware 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 20: 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]



Added system crash alarm, core dump detection and allow users to download core dump file.

[Figure 78: 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 28: SIP Trunk Configuration Parameters]




Added Fail2Ban support for SIP authentication. [FAIL2BAN]



Added voice prompt "Language" selection and "Auto Record" option for extension. [Table 21: SIP

Extension Configuration Parameters] [Table 24: Batch Add SIP Extension Parameters]



Added "Auto Record" option for trunk. [Table 26: Analog Trunk Configuration Parameters] [Table 28:

SIP Trunk Configuration Parameters]



Added support to specify caller ID in outbound route and inbound route. [Table 30: Outbound Route

Configuration Parameters] [Table 31: Inbound Rule Configuration Parameters]



Added "Digit Timeout" option and voice prompt "Language" selection for IVR. [Table 34: IVR




Added "Direct Dial Voicemail Prefix" feature code to directly dial or transfer to extension's voicemail.

[Table 44: UCM6100 Feature Codes]



Added "Enforce Strong Passwords" option. [Table 45: Internal Options/General]



Added FXS MWI Mode. [Table 48: Internal Options/Hardware Config]



Added system events with alert and Email notification support. [SYSTEM EVENTS]



Added new web page for recording files. [RECORDING FILES]






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

20: Auto Provision Settings]



Changed one of the discovery method from "SIP MESSAGE (OPTIONS)" to "SIP MESSAGE

(NOTIFY)" in zero-config feature. [DISCOVERY]



Added pickup group feature. [Table 44: 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 28: SIP Trunk Configuration

Parameters]



Added LDAP sync options for peer SIP trunk. [Table 28: 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 30: Outbound Route Configuration

Parameters]



Added DISA and Fax to inbound route default destination options. [Table 31: Inbound Rule




Added DISA and Fax to IVR key press event options. [Table 34: IVR Configuration Parameters]



Added "Min Message Time" option in voicemail settings. [Table 35: 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 44: UCM6100 Feature Codes]



Added feature code *8 for "Pickup Extension" in pickup group feature. [Table 44: UCM6100 Feature

Codes]



Added "Record Prompt" and "Custom Name of Pickup Group" options in internal options. [Table 45:

Internal Options/General]



Added warning information for "Allow Guest Call" option to avoid potential security risk caused by

misconfiguration. [Table 50: IAX Settings/General]



Changed reset mode to two mode "User Data" and "All". [RESET AND REBOOT]






This is the initial version.



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.



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

2 ports (both with lifetime capability in case of power outage)

PSTN Line FXO Ports



UCM6102: 2 ports



UCM6104: 4 ports



UCM6108: 8 ports



UCM6116: 16 ports

Network Interfaces



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

Peripheral Ports

LED Indicators

Yes, UCM6102 only

USB, SD/SDHC (VFAT)

Power/Ready, Network, PSTN Line, USB, SD



LCD Display

Reset Switch

Voice/Video Capabilities

128x32 graphic LCD with DOWN and OK button

Yes

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

H.264, H.263, H.263+

Layer 3 QoS

Video Codecs

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

Physical

SRTP, TLS, HTTPS, SSH

Universal Power Supply

Environmental

Dimensions

Weight

Mounting



Output: 12VDC, 1.5A



Input: 100-240VAC, 50-60Hz



Operating: 32 - 104 o

F / 0 - 40 o

C, 10-90% (non-condensing)



Storage: 14 - 140 o

F / -10 - 60 o

C



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












UCM6102/6104: Wall mount and Desktop



UCM6108/6116: Rack mount and Desktop



Additional Features

Multi-language Support

Caller ID

Polarity Reversal/ Wink

Yes, English/Chinese/Spanish/French/German/Russian/Italian for Web

GUI; Customizable IVR to support any language

Bellcore/Telcordia, ETSI-FSK, ETSI-DTMF, SIN 227 - BT, NTT Japan

Yes, with enable/disable option upon call establishment and termination

Call Center

Multiple configurable call queues, automatic call distribution (ACD) based on agent skills/availability busy level, in-queue announcement

Customizable Auto Attendant

Up to 5 layers of IVR (Interactive Voice Response)

Concurrent Calls



UCM6102: Up to 30 simultaneous calls



UCM6104: Up to 45 simultaneous calls



UCM6108/6116: Up to 60 simultaneous calls

Conference Bridges



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)



UL 60950 (power adapter)



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

Power Adaptor

Ethernet Cable

Quick Installation Guide

Yes (1)

Yes (1)

Yes (1)

Yes (1)

Table 3: UCM6108/UCM6116 Equipment Packaging

Main Case

Power Adaptor

Ethernet Cable

Quick Installation Guide

Wall Mount

Screws

Yes (1)

Yes (1)

Yes (1)

Yes (1)

Yes (2)

Yes (6)

CONNECT YOUR UCM6100

CONNECT THE UCM6102



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.







1. Connect one end of an RJ-45 Ethernet cable into the LAN 1 port of the UCM6104.




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.







1. Connect one end of an RJ-45 Ethernet cable into the LAN port of the UCM6108.




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.











1. Connect one end of an RJ-45 Ethernet cable into the LAN port of the UCM6116.




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.



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.



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.



View Events

Device Info

Network Info

Network Menu

Factory Menu


Table 4: LCD Menu Options



Critical Events



Other Events



Hardware: Hardware version number



Software: Software version number



P/N: Part number



WAN MAC: WAN side MAC address (UCM6102 only)



LAN MAC: LAN side MAC address



Uptime: System up time

For UCM6104/UCM6108/UCM6116:



LAN Mode: DHCP, Static IP, or PPPoE



LAN IP: IP address



LAN Subnet Mask

For UCM6102:



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

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 patterns. When done, press "OK" button to exit.



Fan Mode

Select "Auto" or "On".



LED Test Patterns


Web Info

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.



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

LAN

WAN

USB

SD

FXS (Phone/Fax)

FXO (Telco Line)

LED Status

Solid: Connected

Flashing: Data Transferring

OFF: Not Connected

LED

NETWORK

Table 6: UCM6108/UCM6116 LED INDICATORS

LED Status

Solid: Connected


ACT

USB

SD

Phone (FXS)

Line (FXO)

Solid: Connected

Flashing: Data Transferring




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



4. Enter the administrato r’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:



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.

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.



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

Method

Select "Route", "Switch" or "Dual" mode on the network interface of UCM6102.

The default setting is "Route".



Route

WAN port interface will be used for uplink connection. LAN port interface will be used to serve as router.



Switch

WAN port interface will be used for uplink connection. LAN port interface will be used as bridge for PC connection.



Dual



Both ports can be used for uplink connection. Users will need assign LAN 1 or LAN 2 as the default interface in option "Default Interface" and configure

"Gateway IP" for this interface.

Preferred DNS Server Enter the preferred DNS server address.

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

IP Method

Gateway IP

Subnet Mask

IP Address

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

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

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.

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

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

0.0.0.0.

Enter the DNS server 2 address for static IP settings.

Enter the user name to connect via PPPoE.

Enter the password to connect via PPPoE.

Assign the VLAN tag of the layer 2 QoS packets for WAN port. The default value is 0.

Layer 2 QoS 802.1p

Priority Value

Assign the priority value of the layer 2 QoS packets for WAN port. The default value is 0.

LAN (when Method is set to "Route")

IP Address Enter the IP address assigned to LAN port. The default setting is 192.168.2.1.

Subnet Mask Enter the subnet mask. The default setting is 255.255.255.0.

DHCP Server Enable Enable or disable DHCP server capability. The default setting is "Yes".

DNS Server 1

DNS Server 2

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

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

Gateway IP

Subnet Mask

IP Address

DNS Server 1




255.255.0.0.





DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

0.0.0.0.




Assign the VLAN tag of the layer 2 QoS packets for LAN port. The default value is 0.

Layer 2 QoS 802.1p

Priority Value

Assign the priority value of the layer 2 QoS packets for LAN port. The default value is 0.

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

Default Interface

IP Method

Gateway IP

IP Address

Subnet Mask

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

If "Dual" is selected as "Method", users will need assign the default interface to 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.


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.



255.255.0.0.


0.0.0.0.






Table 8: UCM6104 Network Settings->Basic Settings

Method

Select "Switch" or "Dual" mode on the network interface of UCM6104. The default setting is "Switch".



Switch

LAN 1 port interface will be used for uplink connection. LAN 2 port interface will be used as bridge for PC connection.



Dual



Both ports can be used for uplink connection. Users will need assign the default interface in option "Default Interface". Users will need assign LAN 1 or LAN 2 as the default interface in option "Default Interface" and configure

"Gateway IP" for this interface.


LAN (when Method is set to "Switch")

IP Method

Gateway IP

Subnet Mask

IP Address




255.255.0.0.


DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag


0.0.0.0.





Layer 2 QoS 802.1p

Priority Value


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

Default Interface

IP Method

Gateway IP

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.



Subnet Mask

IP Address

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p


255.255.0.0.



0.0.0.0.





Assign the priority value of the layer 2 QoS packets for LAN port. The default



Priority Value value is 0.

Table 9: UCM6108/UCM6116 Network Settings->Basic Settings


IP Method

Gateway IP


Enter the gateway IP address for static IP settings.

Subnet Mask

IP Address

DNS Server 1

DNS Server 2

Enter the subnet mask address for static IP settings.

Enter the IP address for static IP settings.



User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value





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

802.1X Mode

Identity

MD5 Password

802.1X Certificate

802.1X Client

Certificate

Select 802.1X mode. The default setting is "Disable". The supported 802.1X mode are:



EAP-MD5



EAP-TLS



EAP-PEAPv0/MSCHAPv2

Enter 802.1X mode identity information.

Enter 802.1X mode MD5 password information.

Select 802.1X certificate from local PC and then upload.

Select 802.1X client certificate from local PC and then upload.



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

LAN IP

LAN Port

Protocol Type

Specify the WAN port number. Up to 8 ports can be configured.

Specify the LAN IP address.

Specify the LAN port number.

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

FIREWALL

The UCM6100 provides users firewall configurations to prevent certain malicious attack to the UCM6100 system. Users could configure to allow, restrict or reject specific traffic through the device for security and bandwidth purpose. The UCM6100 also provides Fail2ban feature for authentication errors in SIP

REGISTER, INVITE and SUBSCRIBE. To configure firewall settings in UCM6100, go to Web

GUI->Settings->Firewall page.

STATIC DEFENSE

Under Web GUI->Settings->Firewall->Static Defense page, users will see the following information:



Current service information with port, process and type.



Typical firewall settings.



Custom firewall settings.

The following table shows a sample current service status running on the UCM6100.



Table 12: UCM6100 Firewall->Static Defense->Current Service

Port

7777

389

22

80

Process

Asterisk

Slapd

Dropbear

Lighthttpd

Type

tcp/IPv4 tcp/IPv4 tcp/IPv4 tcp/IPv4

Protocol or Service

SIP

LDAP

SSH

HTTP

8089

69

9090

6060

Lighthttpd

Opentftpd

Asterisk zero_config tcp/IPv4 udp/IPv4 udp/IPv4 udp/IPv4

HTTPS

TFTP

SIP

UCM6100 zero_config service

5060

4569

5353

37435

Asterisk

Asterisk zero_config

Syslogd udp/IPv4 udp/IPv4 udp/IPv4 udp/IPv4

SIP

SIP

UCM6100 zero_config service

Syslog

For typical firewall settings, users could configure the following options on the UCM6100.

Table 13: Typical Firewall Settings

Ping Defense

Enable

If enabled, ICMP response will not be allowed for Ping request. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN (UCM6102 only) interface.

SYN-Flood Defense

Enable

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.

Ping-of-Death

Defense Enable

Enable to prevent Ping-of-Death attack to the device. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN

(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.



Figure 11: Create New Firewall Rule

Table 14: Firewall Rule Settings

Rule Name

Action

Type

Specify the Firewall rule name to identify the firewall rule.

Select the action for the Firewall to perform.



ACCEPT



REJECT



DROP

Select the traffic type.



IN

If selected, users will need specify the network interface "LAN" or "WAN"

(for UCM6102 only) for the incoming traffic.



OUT

Service

Select the service type.



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 to delete the rule.



DYNAMIC DEFENSE

Dynamic defense is supported on the UCM6102 only. It can blacklist hosts dynamically when the LAN mode is set to "Route" under web GUI->Settings->Network Settings->Basic Settings page. If enabled, the traffic coming into the UCM6102 can be monitored, which helps prevent massive connection attempts or brute force attacks to the device. The blacklist can be created and updated by the UCM6102 firewall, which will then be displayed in the web page. Please refer to the following table for dynamic defense options on the UCM6102.

Table 15: UCM6102 Firewall Dynamic Defense

Dynamic Defense

Enable

Enable dynamic defense. The default setting is disabled.

Periodical Time

Interval

Blacklist Update

Interval

Connection

Threshold

Dynamic Defense

Whitelist

Configure the dynamic defense periodic time interval (in minutes). If the 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 to 59 when dynamic defense is turned on. The default setting is

59.

Configure the blacklist update time interval (in seconds). The default setting is

120.

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.

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.



Table 16: Fail2Ban Settings

Global Settings

Enable Fail2Ban

Banned Duration

Max Retry Duration

MaxRetry

Fail2Ban Whitelist

Enable Fail2Ban. The default setting is disabled. Please make sure both "Enable

Fail2Ban" and "Asterisk Service" are turned on in order to use Fail2Ban for SIP authentication on the UCM6100.

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 ban the host with matching address in this list. Up to 5 addresses can be added into the list.

Local Settings

Asterisk Service

Port

MaxRetry

Enable Asterisk service for Fail2Ban. The default setting is disabled. Please make sure both "Enable Fail2Ban" and "Asterisk Service" are turned on in order to use Fail2Ban for SIP authentication on the UCM6100.

Configure the listening port number for the service. Currently only 5060 (for

UDP) is supported.

Configure the number of authentication failures during "Max Retry Duration" before the host is banned. The default setting is 10. Please make sure this option 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.



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


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



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 to edit the phonebook attributes and contact list (see figure below), or select to delete the phonebook.


Figure 16: Edit LDAP Phonebook


LDAP CLIENT CONFIGURATIONS

The configuration on LDAP client is similar when you use other LDAP servers. Here we provide an example on how to configure the LDAP client on the SIP end points to use the default PBX phonebook.

Assuming the server base dn is "dc=pbx,dc=com", configure the LDAP clients as follows (case insensitive):

Base DN: dc=pbx,dc=com

Login DN: Please leave this field empty

Password: Please leave this field empty

Anonymous: Please enable this option

Filter: (|(CallerIDName=%)(AccountNumber=%))

Port: 389

To configure Grandstream IP phones as the LDAP client, please refer to the following example:

Server Address: The IP address or domain name of the UCM6100

Base DN: dc=pbx,dc=com

User Name: Please leave this field empty

Password: Please leave this field empty

LDAP Name Attribute: CallerIDName Email Department FirstName LastName

LDAP Number Attribute: AccountNumber MobileNumber HomeNumber Fax

LDAP Number Filter: (AccountNumber=%)

LDAP Name Filter: (CallerIDName=%)

LDAP Display Name: AccountNumber CallerIDName

LDAP Version: If existed, please select LDAP Version 3

Port: 389

The following figure shows the configuration information on a Grandstream GXP2200 to successfully use

the LDAP server as configured in

Figure 12: LDAP Server Configurations

.



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.



Table 17: HTTP Server Settings

Redirect From Port 80

Protocol Type

Port

Enable or disable redirect from port 80. On the PBX, the default access protocol is HTTPS and the default port number is 8089. When this option is enabled, the access using HTTP with Port 80 will be redirected to

HTTPS with Port 8089. The default setting is "Enable".

Select HTTP or HTTPS. The default setting is "HTTPS".

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 18: Email Settings

TLS Enable

Type

Domain

Server

Username

Password

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

Emails sent from MTA might be considered as spam by the target

SMTP server.



Client: Submit Emails to the SMTP server. A SMTP server is required and users need login with correct credentials.

Specify the domain name to be used in the Email when using type

"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".



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.

Table 19: Time Auto Updating



Remote NTP Server

Enable DHCP Option 2

Enable DHCP Option 42

Time Zone

Self-Defined Time Zone

Specify the URL or IP address of the NTP server for the UCM6100 to 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 from DHCP Option 2 in the local server automatically. The default setting is "Yes".

If set to "Yes", the UCM6100 is allowed to get provisioned for NTP

Server 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.

If "Self-Defined Tome Zone" is selected, please specify the time zone parameters in "Self-Defined Time Zone" field as described in below option.

If "Self-Defined Time Zone" is selected in "Time Zone" option, users will need define their own time zone following the format below.

The syntax is: std offset dst [offset], start [/time], end [/time]

Default is set to: MTZ+6MDT+5,M4.1.0,M11.1.0

MTZ+6MDT+5

This indicates a time zone with 6 hours offset and 1 hour ahead for DST, which is U.S central time. If it is positive (+), the local time zone is west of the Prime Meridian (A.K.A: International or Greenwich Meridian); If it is 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.



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.



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.

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.



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.



Figure 21: Auto Provision Settings

Table 20: Auto Provision Settings

Enable Zero Config

Enable or disable the zero config feature on the PBX. The default setting is disabled.

Automatically Assign Extension

If enabled, when the device is discovered, the PBX will automatically assign an extension within the range defined in "Zero Config Extension

Segment" to the device. The default setting is disabled.

Zero Config Extension

Segment

Enable Pick Extension

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 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):

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.



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

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 icon for the entry in the zero config device list.

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.



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 21: SIP Extension Configuration Parameters

General

Extension

CallerID Number

Permission

SIP/IAX Password

Enable Voicemail

Voicemail Password

Call Forward Unconditional

The extension number associated with the user.

Configure the CallerID Number that would be applied for outbound calls from this user.

Note:

The ability to manipulate your outbound Caller ID may be limited by your

VoIP provider.

Assign permission level to the user. The available permissions are

"Internal", "Local", "National" and "International" from the lowest level to the highest level. The default setting is "Internal".

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 automatically generated. It is recommended to use this password for security purpose.

Enable voicemail for the user. The default setting is "Yes".

Configure voicemail password (digits only) for the user to access the voicemail box. A random numeric password is automatically generated.

It is recommended to use the random generated password for security purpose.

Configure the Call Forward Unconditional target number. If not configured, the Call Forward Unconditional feature is deactivated. The default setting is deactivated.



Call Forward No Answer

Call Forward Busy

Ring Timeout

Auto Record

Skip Voicemail Password

Verification

Support Hot-Desking Mode

User Settings

First Name

Last Name

Email Address

Language

Configure the Call Forward No Answer target number. If not configured, 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 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 under web GUI->Internal Options->IVR Prompt: General Preference.

The valid range is between 5 seconds and 600 seconds.

Note:

If the end point also has a ring timeout configured, the actual ring timeout used is the shortest time set by either device.

Enable automatic recording for the calls using this extension. The default setting is disabled. The recording files can be accessed under web GUI->CDR->Recording Files.

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

Configure the first name of the user. The first name can contain characters, letters, digits and _.

Configure the last name of the user. The last name can contain characters, letters, digits and _.

Fill in the Email address for the user. Voicemail will be sent to this Email address.

Select the voice prompt language to be used for this extension. The default setting is "Default" which is the selected voice prompt language under web GUI->PBX->Internal Options->Language. The dropdown list 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

NAT

Can Reinvite

DTMF Mode

Insecure

Enable Keep-alive

Keep-alive Frequency

Auth ID

Other Settings

SRTP

Fax Detection


Use NAT when the UCM6100 is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it's related to NAT configuration or Firewall's support of SIP and RTP ports. The default setting is enabled.

By default, the UCM6100 will route the media steams from SIP endpoints through itself. If enabled, the PBX will attempt to negotiate with the 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

"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 required.



No: Normal IP-based peers matching and authentication of incoming INVITE.

The default setting is "Port".

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

CallerID number

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 default Email address in Fax setting page under web

GUI->PBX->Internal Options->Fax/T.38.


Strategy

Skip Trunk Auth

Codec Preference

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 three subnet addresses can be specified.



A Specific IP Address

Only the device on the specific IP address can register this extension.

The default setting is "Allow All".

If enabled, users will not need enter the "PIN Set" required by the outbound rule to make outbound calls. The default setting is "No".

Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729,

G.723, ILBC, 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.

Table 22: IAX Extension Configuration Parameters

General

Extension

CallerID Number

Permission



Note:


VoIP provider.


"Internal", "Local", "National" and "International" from the lowest level to



SIP/IAX Password

Enable Voicemail

Voicemail Password



Call Forward Busy

Ring Timeout

Auto Record


Verification the highest level. The default setting is "Internal".

Note:


Configure the password for the user. A random secure password will be automatically generated. It is recommended to use this password for security purpose.











Note:






User Settings

First Name

Last Name

Email Address

Language

IAX Settings

Max Number of Calls

Require Call Token

Other Settings

SRTP

Fax Detection

Strategy








Configure the maximum number of calls allowed for each remote IP address.

Configure to enable/disable requiring call token. If set to "Auto", it might lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints. The default setting is "Yes".

Enable SRTP for the call. The default setting is disabled.




Note:





Allow All




Local Subnet Only



Skip Trunk Auth

Codec Preference



A Specific IP Address




Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G,726, G.722, G.729,

G.723, ILBC, 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 23: FXS Extension Configuration Parameters

General

Extension

Analog Station

CallerID Number

Permission

Enable Voicemail

Voicemail Password


Select the FXS port to be assigned for this extension.


Note:


VoIP provider.



Note:








Call Forward Busy

Ring Timeout

Auto Record









Note:





Verification

User Settings

First Name

Last Name

Email Address

Language







Analog Settings

Call Waiting

User # as SEND

RX Gain

TX Gain

MIN RX Flash

MAX RX Flash

Enable Polarity Reversal

Echo Cancellation

3-Way Calling

Send CallerID After

Other Settings

Fax Detection




Configure to enable/disable call waiting feature. The default setting is

"No".

If configured, the # key can be used as SNED key after dialing the number on the analog phone. The default setting is "Yes".

Configure the RX gain for the receiving channel of analog FXS port. The valid range is -30dB to +6dB. The default setting is 0.

Configure the TX gain for the transmitting channel of analog FXS port.

The valid range is -30dB to +6dB. The default setting is 0.

Configure the minimum period of time (in milliseconds) that the hook-flash must remain unpressed for the PBX to consider the event as a valid flash event. The valid range is 30ms to 1000ms. The default setting is 200ms.

Configure the maximum period of time (in milliseconds) that the hook-flash must remain unpressed for the PBX to consider the event as a valid flash event. The minimum period of time is 256ms and it can't be modified. The default setting is 1250ms.

If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity 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.

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.

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.

Enable to detect Fax signal from the user/trunk during the call and send


Skip Trunk Auth the received Fax to the Email address configured for this extension. If no



Note:



BATCH ADD EXTENSIONS

BATCH ADD SIP EXTENSIONS

Under Web GUI->PBX->Basic/Call Routes->Extensions, click on "Batch Add Extensions"->"Batch Add

SIP Extensions".

Table 24: Batch Add SIP Extension Parameters

General

Start Extension

Create Number

Permission

Enable Voicemail

SIP/IAX Password

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.



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 for the user. The default setting is "Yes".

Configure the SIP/IAX password for the users. Three options are available to create password for the batch of extensions.



User Random Password.

A random secure password will be automatically generated. It is recommended to use this password for security purpose.



Use Extension as Password.



Voicemail Password

Ring Timeout

Auto Record


Verification

SIP Settings

NAT

Can Reinvite

DTMF Mode



Enter a password to be used on all the extensions in the batch.

Configure Voicemail password (digits only) for the users.




A random password in digits will be automatically generated. It is recommended to use this password for security purpose.










Note:




Use NAT when the PBX is on a public IP communicating with devices hidden behind NAT (e.g., broadband router). If there is one-way audio issue, usually it'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 endpoints to route the media stream directly. It is not always possible for the PBX to negotiate endpoint-to-endpoint media routing. The default setting is "No".

Select DTMF mode for the user to send DTMF. The default setting is

"RFC2833". If "Info" is selected, SIP INFO message will be used. If

"Inband" is selected, 64-kbit codec PCMU and PCMA are required.

When "Auto" is selected, RFC2833 will be used if offered, otherwise

"Inband" will be used.



Insecure

Enable Keep-alive

Keep-alive Frequency

Other Settings

SRTP

Fax Detection

Strategy

Skip Trunk Auth

Codec Preference




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 required.



No: Normal IP-based peers matching and authentication of incoming INVITE.

The default setting is "Port".

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.


Enable SRTP for the call. The default setting is "No".




Note:





Allow All




Local Subnet Only




A Specific IP Address.




Select audio and video codec for the extension. The available codecs 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".

Table 25: Batch Add IAX Extension Parameters

General

Start Extension

Create Number

Permission

Enable Voicemail

SIP/IAX Password

Voicemail Password

Ring Timeout

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.



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 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.




A random secure password will be automatically generated. It is recommended to use this password for security purpose.







Configure Voicemail password (digits only) for the users.




A random password in digits will be automatically generated. It is recommended to use this password for security purpose.








UCM6100, which can be configured in the global ring timeout setting



Auto Record


Verification

IAX Settings

Max Number of Calls

Require Call Token

Other Settings

SRTP

Fax Detection

Strategy under web GUI->Internal Options->IVR Prompt: General Preference.


Note:




Configure the maximum number of calls allowed for each remote IP address.

Configure to enable/disable requiring call token. If set to "Auto", it might lock out users who depend on backward compatibility when peer authentication credentials are shared between physical endpoints. The default setting is "Yes".

Enable SRTP for the call. The default setting is "No".




Note:





Allow All




Local Subnet Only




A Specific IP Address.

Only the device on the specific IP address can register this



Skip Trunk Auth

Codec Preference extension.



Select audio and video codec for the extension. The available codecs are: PCMU, PCMA, GSM, AAL2-G.726-32, G.722, G.729, G.723, ILBC,

ADPCM, LPC10, H.264, H.263 and H.263p.

EDIT EXTENSION

All the UCM6100 extensions are listed under Web GUI->PBX->Basic/Call Routes->Extensions, with status, Extension, CallerID Name, Technology (SIP, IAX and FXS), IP and Port. Each extension has a checkbox for users to "Modify Selected Extensions" or "Delete Selected Extensions". Also, options "Edit"

, "Reboot" and "Delete" are available per extension.



Status

Users can see the following icon for each extension to indicate the SIP status.

Green: Free

Blue: Ringing

Yellow: In Use

Grey: Unavailable



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).



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 26: Analog Trunk Configuration Parameters

Channels

Trunk Name

Select the channel for the analog trunk.



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

Enable Polarity Reversal

Polarity on Answer Delay

Current Disconnect Threshold

(ms)

If enabled, a polarity reversal will be marked as received when an outgoing call is answered by the remote party. For some countries, a polarity 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 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.

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.



Ring Timeout

RX Gain

TX Gain

Use CallerID

Fax Detection

Caller ID Scheme

Auto Record

Tone Settings

Busy Detection

Busy Tone Count

Congestion Detection

Congestion Count

Tone Country

Busy Tone


Configure the ring timeout (in ms). Trunk (FXO) devices must have a timeout to determine if there was a hangup before the line is answered.

This value can be used to configure how long it takes before the

UCM6100 considers a non-ringing line with hangup activity. The default setting is 8000.

Configure the RX gain for the receiving channel of analog FXO port. The valid range 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


Note:


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 setting is disabled. The recording files can be accessed under web

GUI->CDR->Recording Files.

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 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 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 manually configure the values for Busy Tone and Congestion Tone. The default setting is "United States of America (USA)".

Syntax: f1=val[@level][,f2=val[@level]],c=on1/off1[-on2/off2[-on3/off3]];


Congestion Tone

PSTN Detection

Frequencies are in Hz and cadence on and off are in ms.

Frequencies Range: [0, 4000)

Busy Level Range: (-300, 0)

Cadence Range: [0, 16383].

Select Tone Country "Custom" to manually configure Busy Tone value.

Default value: f1=480@-50,f2=620@-50,c=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)

Busy Level Range: (-300, 0)

Cadence Range: [0, 16383].

Select Tone Country "Custom" to manually configure Busy Tone value.

Default value: f1=480@-50,f2=620@-50,c=250/250

Click on "Detect" to detect the busy tone, Polarity Reversal and Current

Disconnect by PSTN. Before the detecting, please make sure there 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.



Figure 26: UCM6100 FXO Tone Settings

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

Figure 27: 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.



Figure 28: UCM6100 PSTN Detection: Auto Detect



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

Figure 29: 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.



Detect Model

Table 27: 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.



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".

Select the channel to be detected.

Select the channel to help detect when "Auto Detect" is used.

Configure the number to be called to help the detection.

Source Channel

Destination Channel

Destination Number

Note:



The PSTN detection process will keep the call up for about 1 minute.



If "Semi-auto Detect' is used, please pick up the call only after informed from the web GUI prompt.



Once the detection is successful, the detected parameters "Busy Tone", "Polarity Reversal" and

"Current Disconnect by PSTN" will be filled into the corresponding fields in the analog trunk configuration.

VOIP TRUNKS

VoIP trunks can be configured in UCM6100 under Web GUI->PBX->Basic/Call Routes->VoIP Trunks.

Once created, the VoIP trunks will be listed with Provider Name, Type, Hostname/IP, Username and

Options to edit/detect the trunk.





Click on "Create New SIP Trunk" or "Create New IAX Trunk" to add a new VoIP trunk.



Click on to configure detailed parameters for the VoIP trunk.



Click on



Click on

to configure Direct Outward Dialing (DOD) for the SIP Trunk.

to start LDAP Sync.



Click on to delete the VoIP trunk.

The VoIP trunk options are listed in the table below.

Table 28: SIP Trunk Configuration Parameters

Create New SIP Trunk

Type

Provider Name

Host Name

Keep Trunk CID

Select the VoIP trunk type.



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 providers 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".

Username

Password

Auth ID

Outbound Proxy

Auto Record

Enter the username to register to the trunk from the provider when

"Register SIP Trunk" type is selected.

Enter the password to register to the trunk from the provider when

"Register SIP Trunk" is selected.

Enter the Authentication ID for "Register Trunk" type.

Enter the IP address or URL of the outbound proxy for "Register SIP

Trunk" type.

Enable automatic recording for the calls using this trunk (for SIP trunk only). The default setting is disabled. The recording files can be accessed 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



Transport

Keep Trunk CID

Caller ID

CallerID Name

Codec Preference

Auto Record

DID Mode

Enable Qualify


trunk.

Configure the SIP transport protocol to be used in this trunk. The default setting is "All - UDP Primary".



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.


Configure the Caller ID. This is the number that the trunk will try to use when making outbound calls. For some providers, it might not be possible to set the CallerID with this option and this option will be ignored.

When making outgoing calls, the following rules are used to determine which CallerID will be used if they exist:



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.

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 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 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".


Qualify Timeout

Qualify Frequency

Fax Detection

SRTP

Sync LDAP Enable

Sync LDAP Password

Sync LDAP Port

When "Enable Qualify" option is set to "Yes", configure the timeout (in 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 seconds) of the SIP OPTIONS message sent to the device to check if the device is still online. The default setting is 60 seconds.



Note:


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 ensure successful synchronization, the remote UCM6100 peer also needs to enable this option on the SIP peer trunk. The default setting is

"No".

This is the password used for LDAP contact file encryption and decryption during the LDAP sync process. The password must be the same on both UCM6100 peers o ensure successful synchronization.

Configure the TCP port used LDAP sync feature between two peer

UCM6100.

LDAP Outbound Rule

Specify an outbound rule for LDAP sync feature. The UCM6100 will 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

Transport


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".



UDP Only



TCP Only



TLS Only



Keep Trunk CID

Username

Password

Auth ID

Codec Preference

From Domain

From User

Outbound Proxy Support

Outbound Proxy

Auto Record

DID Mode

Enable Qualify

Qualify Timeout




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.

When enabled, it can avoid overridden by extension's CID if the extension has CID configured. The default setting is enabled.

Enter the username to register to the trunk from the provider.

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 required by the provider. If not specified, the CallerID name will be sued for authentication.


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.

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 for registration (single trunk) with multiple DIDs.

For example, "1234567" is the From User in From Header: sip:[email protected].

Select to enable outbound proxy in this trunk. The default setting is "No".

When outbound proxy support is enabled, enter the IP address or URL of the outbound proxy.



Configure where to get the destination ID of an incoming SIP call, from

SIP Request-line or To-header. The default is set to "Request-line".


When "Enable Qualify" option is set to "Yes", configure the timeout (in


Qualify Frequency

Fax Detection ms) for the Qualify SIP message. If no response is received within the timeout, the device is considered offline. The default setting is 1000ms.




Note:


Enable SRTP for the VoIP trunk. The default setting is "No". SRTP

Table 29: IAX Trunk Configuration Parameters

Create New IAX Trunk

Type

Provider Name

Host Name

Keep Trunk CID

Select the VoIP trunk type.



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 providers server of the trunk.


Username

Password

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.

Peer IAX Trunk Configuration Parameters

Provider Name

Host Name

Keep Trunk CID

Caller ID




Configure the Caller ID. This is the number that the trunk will try to use



CallerID Name

Codec Preference

Enable Qualify 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.














G.723, ILBC, ADPCM, H.264, H.263, H.263p.


Qualify Timeout

Qualify Frequency

Host Name

Keep Trunk CID



Fax Detection



Note:


Register IAX Trunk Configuration Parameters

Provider Name



When enabled, it can avoid overridden by extension's CID if the extension has CID configured. The default setting is enabled.



Caller ID

CallerID Name

Username

Password

Auth ID

Codec Preference

Enable Qualify

Qualify Timeout

Qualify Frequency

Fax Detection


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.













Enter the username to register to the trunk from the provider.

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 required by the provider. If not specified, the CallerID name will be sued for authentication.


G.723, ILBC, ADPCM, H.264, H.263, H.263p.






Note:



Direct Outward Dialing (DOD)

The UCM6100 provides Direct Outward Dialing (DOD) which is a service of a local phone company (or local exchange carrier) that allows subscribers within a company's PBX system to connect to outside lines directly.

Example of how DOD is used:

Company ABC has a SIP trunk. This SIP trunk has 4 DIDs associated to it. The main number of the office is routed to an auto attendant. The other three numbers are direct lines to specific users of the company.

At the moment when a user makes an outbound call their caller ID shows up as the main office number.

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 30: DOD extension selection


6. Click "Save" at the bottom.

Once completed, the user will return to the EDIT DOD page that shows all the extensions that are associated to a particular DOD.

Figure 31: Edit DOD



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 30: Outbound Route Configuration Parameters

Calling Rule Name

Pattern

Password

Configure the name of the calling rule (e.g., local, long_distance, and etc). Letters, digits, _ and - are allowed.



All patterns are prefixed with the "_".



Special characters:

X: Any Digit from 0-9.

Z: Any Digit from 1-9.

N: Any Digit from 2-9.

".": Wildcard. Match one or more characters.

"!": Wildcard. Match zero or more characters immediately.

Example: [12345-9] - Any digit from 1 to 9.



To specify the outbound route based on caller ID, add "/callerID" to the above pattern. The outbound call has to match both the pattern and the callerID to use this outbound route.

Example: _5XXX/1234567890

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



Privilege Level

Enable Filter on Source Caller

ID

Strip

Prepend calls.

Select privilege level for the outbound rule.

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



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



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



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

The default setting is "International". 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 select extensions allowed to use this outbound route. "Privilege Level" is automatically disabled if using

"Enable Filter on Source Caller ID".

Custom Dynamic Route

If "Enable Filter on Source Caller ID" is turned on, users could also define the pattern for the source caller ID allowed to use this outbound route, besides specifying single extensions.






Special characters:







Send This Call Through Trunk

Use Trunk Select the trunk for this outbound rule.

Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk.

Example:

The users will dial 9 as the first digit of a long distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed.

Specify the digits to be prepended before the call is placed via the trunk.



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

Use Failover Trunk

Failover Trunk

Strip

Prepend

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.

Example:

The user's primary trunk is a VoIP trunk and the user would like to use the PSTN when the VoIP trunk is not available. The PSTN trunk can be configured as the failover trunk of the VoIP trunk.

Allows the user to specify the number of digits that will be stripped from the beginning of the dialed string before the call is placed via the selected trunk.

Example:

The users will dial 9 as the first digit of a long distance calls. However, 9 should not be sent out via analog lines and the PSTN line. In this case, 1 digit should be stripped before the call is placed.

Specify the digits to be prepended before the call is placed via the trunk.

Those 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.

INBOUND RULE CONFIGURATIONS

Table 31: Inbound Rule Configuration Parameters

Trunks

DID Pattern

Select the trunk to configure the inbound rule.






Privilege Level

Default Destination




Special characters:









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.



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 compared 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.



Extension



Voicemail



Conference Room



Queue



Ringgroup



Page



Vmgroup



Fax



DISA



IVR



By DID

When "By DID" is used, the UCM6100 will look for the destination


DID Destination

Time Condition

Start Time

End Time

Date

Week

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.

Select the DID destination. Only the selected category can be reached by DID using this inbound route.



Extension



Conference



Call Queue



Ring Group



Paging/Intercom Group



IVR



Voicemail Groups



Fax Extension

Select the start time "hour:minute" for the trunk to use the inbound rule.

Select the end time "hour:minute" for the trunk to use the inbound rule.

Select "By Week" or "By Day" and specify the date for the trunk to use the inbound rule.

Select the day in the week to use the inbound rule.

Select the default destination for the inbound call.



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



Ringgroup



Page



Vmgroup



Fax



DISA



IVR


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 32: Blacklist Configuration Parameters



Select the checkbox for "Blacklist Enable" to turn on Blacklist feature for all inbound routes. Blacklist is disabled by default.



Enter a number in "Add Blacklist Number" field and then click to add to the list.



To remove a number from the Blacklist, select the number in "Blacklist list" and click on .

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.



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.

Extension

Password

Admin Password

Table 32: Conference Bridge Configuration Parameters

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.

Note:



If "Public Mode" is enabled, the password is not required to join the conference bridge thus this field is invalid.



The password has to be at least 4 characters.

Configure the password to join the conference bridge as administrator.

Conference administrator can manage the conference call via IVR (if

"Enable Caller Menu" is enabled) as well as invite other parties to join the conference by dialing "0" (permission required from the invited party) or "1" (permission not required from the invited party) during the



Enable Caller Menu

Record Conference

Quiet Mode

Wait For Admin

Enable User Invite

Announce Callers

Public Mode


conference call.

Note:



If "Public Mode" is enabled, the password is not required to join the conference bridge thus this field is invalid.




If enabled, conference participant could press the * key to access the conference bridge menu. The default setting is "No".

If enabled, the calls in this conference bridge will be recorded automatically 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".

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".

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".

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".

Note:

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

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


Play Hold Music For First

Caller

Skip Authentication When

Inviting User via Trunk from

Web GUI

Music On Hold

If enabled, the UCM6100 will play Hold music to the first participant in the conference until another user joins in. The default setting is "No".

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

Select the music on hold class to be played in conference call. Music On

Hold class can be set up under web UI->PBX->Internal

Options->Music On Hold.

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 33: Conference Invitation From Web GUI



Invite by dialing 0 or 1 during conference call.



A conference participant can invite other parties to the conference by dialing from the phone during the conference call. Please make sure option "Enable User Invite" is turned on for the conference bridge first.

Enter 0 or 1 during the conference call. Follow the voice prompt to input the number of the party you would like to invite. A call will be sent to this number to join it into the conference.

0: If 0 is entered to invite other party, once the invited party picks up the invitation call, a permission will be asked to "accept" or "reject" the invitation before joining the conference.

1: If 1 is entered to invite other party, no permission will be required from the invited party.

Note:

Conference administrator can always invite other parties from the phone during the call by entering 0 or 1.

To join a conference bridge as administrator, enter the admin password when joining the conference. A conference bridge can have multiple administrators.

DURING THE CONFERENCE

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



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.



Table 33: Conference Caller IVR Menu

1

4

5

6

7

8

Conference Administrator IVR Menu

1 Mute/unmute yourself.

2

3

4

Lock/unlock the conference bridge.

Kick the last joined user from the conference.

Decrease the volume of the conference call.

5

6

7

8

Decrease your volume .

Increase the volume of the conference call.

Increase your volume.

More options.



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

Mute/unmute yourself.

Decrease the volume of the conference call.

Decrease your volume.

Increase the volume of the conference call.

Increase your volume.

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.



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 to download the recording or click on to delete the recording.

Figure 34: Conference Recording



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.

Name

Extension

Dial Other Extensions

Dial Trunk

Permission

Welcome Prompt

Digit Timeout

Response Timeout

Table 34: IVR Configuration Parameters

Configure the name of the IVR. Letters, digits, _ and - are allowed.

Enter the extension number for users to access the IVR.

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 must be configured for the users to use the trunk first. The default setting is "No".

Assign permission level for outbound calls if "Dial Trunk" is enabled. The available permissions are "Internal", "Local", "National" and

"International" from the lowest level to the highest level. The default setting is "Internal". If the user tries to dial outbound calls after dialing into the IVR, the 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

"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, 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.

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



Response Timeout Prompt

Invalid Prompt

Response Timeout Repeat

Loops

Invalid Repeat Loops

Language

Key Press Event detected within the timeout, a timeout prompt will be played. The default setting is 10 seconds.

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. When the loop ends, it will go to the invalid destination if configured, or hang up. The default setting is 3.

Select the voice prompt language to be used for this IVR. The default setting is "Default" which is the selected voice prompt language under web GUI->PBX->Internal Options->Language. The dropdown list shows all the current available voice prompt languages on the



Select the event for each key pressing for 0-9, *, Timeout and Invalid.

The event options are:



Extension



Voicemail



Conference Rooms



Voicemail Group



IVR



Ring Group



Queues



Page Group



Fax



IVR Prompt



Hangup



DISA

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 35: 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 36: 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.



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 37: 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.



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.



Figure 38: 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 39: Voice Prompt Package List

Click on to download the language to the UCM6100. The installation will be automatically started once the downloading is finished.



Figure 40: 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



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 35: Voicemail Settings

Max Greeting

Dial '0' For Operator

Max Messages Per Folder

Max Message Time

Configure the maximum number of seconds for the voicemail greeting.


If enabled, the caller can press 0 to exit the voicemail application and connect to the configured operator's extension. The operator extension can be configured under web GUI->PBX->Internal Options->General.

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:



1 minute



2 minutes



5 minutes



15 minutes



30 minutes



Unlimited

Min Message Time

Configure the minimum duration (in seconds) of a voicemail message.

Messages will be automatically deleted if the duration is shorter than the

Min Message Time. The default setting is 3 seconds. The available options are:



No minimum



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 If enabled, the caller ID of the user who has left the message will be



Announce Message Duration

Play Envelope

Allow User Review announced at the beginning of the voicemail message. The default setting is "No".

If enabled, the message duration will be announced at the beginning of the voicemail message. The default setting is "No".

If enabled, a brief introduction (received time, received from, and etc) of each message will be played when accessed from the voicemail application. The default setting is "Yes".

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 41: Voicemail Email Settings



Table 36: Voicemail Email Settings

Attach Recordings to E-Mail

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

Template For Voicemail Emails

Fill in the "Subject:" and "Message:" content, to be used in the Email when sending to the user.

The template variables are:



\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 42: Voicemail Group


Voicemail Group Extension

Name

Voicemail Group Mailboxes

Table 37: Voicemail Group Settings

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.

Select available mailboxes from the left list and add them to the right list.

The extensions need to have voicemail enabled to be listed in available mailboxes list.



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 43: 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 38: Ring Group Parameters

Ring Group Name

Extension

Configure ring group name to identify the ring group. Letters, digits, _ and - are allowed.

Configure the ring group extension.

Ring Group Members

Ring Strategy

Select available users from the left side to the ring group member list on the right side. Click on to arrange the order.

Select the ring strategy. The default setting is "Ring in order".



Ring simultaneously.

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

Ring Timeout on Each Member

Configure the number of seconds to ring each member. If set to 0, it will keep ringing. The default setting is 30 seconds.



Enable Destination

Secret

Email Address

Note:

The actual ring timeout might be overridden by users if the phone has ring timeout settings as well.

If enabled, users could select extension, voicemail, ring group, IVR, call queue, voicemail group as the destination if the call to the ring group has no answer. Secret and Email address are required if voicemail is selected as the destination.

Configure the password to access the ring group extension's voicemail.

Note:


Configure the Email address of the ring group extension's voicemail. If

"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 44: Ring Group Configuration



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.

Name

Extension

Type

Page/Intercom Group

Members



Click on to edit the paging/intercom group.

Figure 45: Paging/Intercom Group

Table 39: Paging/Intercom Group Configuration Parameters

Configure paging/intercom group name.

Configure the paging/intercom group extension.

Select "2-way Intercom" or "1-way Page".

Select available users from the left side to the paging/intercom group member list on the right.





Click on to delete the paging/intercom group.



Click on "Paging/Intercom Group Settings" to edit Alert-Info Header. This header will be included in the

SIP INVITE message sent to the callee in paging/intercom call.

Figure 46: 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.



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 47: Call Queue



Click on to edit the call queue. The call queue configuration parameters are listed in the table below.

Extension

Name

Strategy

Table 40: Call Queue Configuration Parameters

Configure the call queue extension.

Configure the call queue name to identify the call queue.

Select the strategy for the call queue.



Ring All

Ring all available Agents simultaneously until one answers.



Linear

Ring agents in the specified order.



Least Recent

Ring the agent who has been called the least recently.



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.



Music On Hold

Leave When Empty

Dial in Empty Queue

Dynamic Login Password

Ring Time Out

Wrapup Time

Max Queue Length


The default setting is "Ring All".

Select the Music On Hold class for the call queue.

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.



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.



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.

If enabled, the configured PIN number is required for dynamic agent to log in. The default setting is disabled.

Configure the number of seconds an agent will ring before the call goes to the next agent. The default setting is 15 seconds.

Configure the number of seconds before a new call can ring the queue after 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 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.


Report Hold Time

Wait Time

Agents

If enabled, the UCM6100 will report (to the agent) the duration of time of the call before the caller is connected to the agent. The default setting is

"No".

If enabled, users will be disconnected after the configured number of seconds. The default setting is "No".

Note:

It is recommended to configure "Wait Time" longer than the "Wrapup

Time".

Select the available users to be the static agents in the call queue.

Choose from the available users on the left to the static agents list on the right. Click on to arrange the order.



Click on to delete the call queue.



Click on "Agent Login Settings" to configure Agent Login Extension Postfix and Agent Logout

Extension Postfix. Once configured, users could log in the call queue as dynamic agent.

Figure 48: Agent Login Settings

For example, if the call queue extension is 6500, Agent Login Extension Postfix is * and Agent Logout

Extension Postfix is **, users could dial 6500* to login to the call queue as dynamic agent and dial

6500** to logout from the call queue. Dynamic agent doesn't need to be listed as static agent and can log in/log out at any time.



Call queue feature code "Agent Pause" and "Agent Unpause" can be configured under Web

GUI->PBX->Internal Options->Feature Codes. The default feature code is *83 for "Agent Pause" and *84 for "Agent Unpause".



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 49: Edit Extension Group



Click on to delete the extension group.

USE EXTENSION GROUPS



Here is an example where the extension group can be used. Go to Web GUI->PBX->Basic/Call

Routes->Outbound Routes and select "Enable Filter on Source Caller ID". Both single extensions and extension groups will show up for users to select.

Figure 50: Select Extension Group in Outbound Route



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 51: Edit Pickup Group



Click on to delete the pickup group.



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 52: 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.



FAX/T.38

The UCM6100 supports T.30/T.38 Fax and Fax Pass-through. It can also convert the received Fax to PDF format and send it to the configured Email address. Fax/T.38 settings can be accessed via Web

GUI->PBX->Internal Options->FAX/T.38.

CONFIGURE FAX/T.38



Click on "Create New Fax Extension". In the popped up window, fill the extension, name and Email address to send the received Fax to.



Click on "Fax Settings" to configure the Fax parameters.

Table 41: FAX/T.38 Settings

Enable Error Correction Mode

Maximum Transfer Rate

Minimum Transfer Rate

Default Email Address

Template Variables

Configure to enable Error Correction Mode (ECM) for the Fax. The default setting is "Yes".

Configure the maximum transfer rate during the Fax rate negotiation.

The 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 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.

Note:

The extension's Email address or the Fax's default Email address needs to be configured in order to receive Fax from Email. If neither of them is configured, Fax will be not be received from Email.

Fill in the "Subject:" and "Message:" content, to be used in the Email when sending the Fax to the users.

The template variables are:



${CALLERIDNUM} : Caller ID Number



${CALLERIDNAME} : Caller ID Name



${RECEIVEEXTEN} : The extension to receive the Fax



${FAXPAGES} : Number of pages in the Fax



${VM_DATE} : The date and time when the Fax is received





Click on to edit the Fax extension.



Click on to delete the Fax extension.

SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE

The following instructions 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 53: 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.



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 54: 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 55: Configure Inbound Rule For Fax


Now the Fax configuration is done. When there is an incoming Fax calling to the PSTN number for the

FXO port, it will send the Fax to the Fax machine.

SAMPLE CONFIGURATION FOR FAX-TO-EMAIL

The following instructions 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 56: 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 "Yes".


Figure 57: Enable Fax Detection In Analog Trunk


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 58: Inbound Route To Fax Extension

5. Once successfully configured, the incoming Fax from external Fax machine to the PSTN line number will be converted to PDF file and sent to the Email address [email protected] as attachment.



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.

Name

Password

Permission


Figure 59: Create New DISA

Table 42: DISA Settings

Configure DISA name to identify the DISA.

Configure the password (digit only) required for the user to enter before using DISA to dial out.

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 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


Response Timeout

Digit Timeout route's privilege level, the call will be allowed to go through.

Configure the maximum amount of time the UCM6100 will wait before hanging up if the user dials an incomplete or invalid number. The default setting is 10 seconds.

Configure the maximum amount of time permitted between digits when the user is typing the extension. The default setting is 5 seconds.

Allow Hangup

If enabled, during an active call, users can enter the UCM6100 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.



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 or remote extensions on the peer SIP 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.



Figure 60: Create New Event List

Table 43: Event List Settings

URI

Local Extensions

Remote Extensions

Special Extensions

Configure the name of this event list (for example, office_event_list).

Please note the URI name cannot be the same as the extension name on the UCM6100. The valid characters are letters, digits, _ and -.

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, 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 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



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.



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 44: UCM6100 Feature Codes

Feature Maps

Blind Transfer

Attended Transfer

Disconnect



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.



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.



Options

Disable






Default code: *0.



Enter the code during active call. It will disconnect the call.



Options

Disable



Allow Both: Enable the feature code on both caller and



Call Park

Audio Mix Record callee.



Default code: #72.



Enter the code during active call to park the call.



Options

Disable






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.



Options

Disable




DND/Call Forward

Do Not Disturb (DND) Activate



Default code: *77.

Do Not Disturb (DND) Deactivate



Default code: *78.

Call Forward Busy Activate



Default Code: *90.



Enter the code and follow the voice prompt. Or enter the code followed by the extension to forward the call.

Call Forward Busy Deactivate

Call Forward No Answer Activate



Default Code: *91.



Default Code: *92.





Deactivate



Default Code: *93.


Activate



Default Code: *72.





Deactivate

Feature Misc

Feature Code Digits Timeout





Default Code: *73.

Default Setting: 1000.



Call Park

Parked Lots

Parking Timeout (s)

Feature Codes

Voicemail Access Code

My Voicemail

Agent Pause

Agent Unpause

Paging Prefix

Intercom Prefix

Blacklist Add

Blacklist Remove

Call Pickup on Ringing




Configure the maximum interval (in milliseconds) between the digits input to activate the feature code.



Default Extension: 700.



During an active call, initiate blind transfer and then enter this code to park the call.



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.



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.



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.



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.



Default Code: *80.



To intercom an extension, enter the code followed by the extension number.



Default Code: *40.



To add a number to blacklist for inbound route, dial *40 and follow the voice prompt to enter the number.



Default Code: *41.



To remove a number from current blacklist for inbound route, dial *41 and follow the voice prompt to remove the number.



Default Code: **.



To pick up a call for any extension xxxx, enter the code followed by the extension number xxxx.


Pickup Extension

Direct Dial Voicemail Prefix



Default Code: *8.



This code is for the pickup group which can be assigned for each extension on the extension configuration page.



If there is an incoming call to an extension, the other extensions within the same pickup group can dial *8 directly to pick up the call.



Default Code: *



This code is for the user to directly dial or transfer to an extension's voicemail.



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 extension or trunk, the call will be automatically recorded when there is established call with the extension or trunk.

Otherwise, please follow the instructions below to manually record the call.

1. Make sure the feature code for "Audio Mix Record" is configured and enabled.

2. After establishing the call, enter the "Audio Mix Record" feature code (by default 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 to play the recording or click on to download the recording file.


Figure 61: Download Recording File From CDR Page


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.



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, hardware config and STUN monitor. The options can be accessed via Web GUI->PBX->Internal

Options.

INTERNAL OPTIONS/GENERAL

Table 45: Internal Options/General

General Preferences

Global OutBound CID

Global OutBound CID Name

Operator Extension

Ring Timeout

Record Prompt

Configure the global CallerID used for all outbound calls when no other

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 configured, all outbound calls will have the CallerID Name set to this name. If not, the extension's CallerID Name will be used.

Specify the operator extension, which will be dialed when users presses

0 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.

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 stopped. For example, before recording, the UCM6100 will play voice prompt "The call will be recorded". The default setting is "No".

Extension Preferences

Enforce Strong Passwords

If enabled, strong password will be enforced for the password created on the UCM6100. The default setting is enabled.

Strong Password Rules:

1. Password for voicemail, voicemail group, outbound route, DISA, password, call queue and conference requires non-repetitive and



Enable Random Password

Disable Extension Range non-sequential digits, with a minimum length of 4 digits. Repetitive digits pattern (such as 0000, 1111, 1234, 2345, and etc), or common digits pattern (such as 111222, 321321 and etc) are not allowed to be configured as password.

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 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:



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



Table 46: Internal Options/Jitter Buffer

SIP Jitter Buffer

Enable Jitter Buffer

Jitter Buffer Size

Max Jitter Buffer

Implementation

Select to enable jitter buffer on the sending side of the SIP channel. The default setting is "No".

Configure the time (in ms) to buffer. This is the jitter buffer size used in

"Fixed" jitter buffer, 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 implementation, or used as the jitter buffer size for "Fixed" jitter buffer implementation. The default setting is 200.

Configure the jitter buffer implementation on the sending side of a SIP channel. The default setting is "Fixed".



Fixed

The size is always equal to the value of "Max Jitter Buffer".



Adaptive

The size is adjusted automatically and the maximum value equals to the value of "Max Jitter Buffer".

INTERNAL OPTIONS/RTP SETTINGS

Table 47: Internal Options/RTP Settings

RTP Start

RTP End

Strict RTP

RTP Checksums

Configure the RTP port starting number. The default setting is 10000.

Configure the RTP port ending address. The default setting is 20000.

Configure to enable or disable strict RTP protection. If enabled, RTP packets that do not come from the source of the RTP stream will be dropped. The default setting is "Disable".

Configure to enable or disable RTP Checksums on RTP traffic. The default setting is "Disable".

INTERNAL OPTIONS/HARDWARE CONFIG

The analog hardware (FXS port and FXO port) on the UCM6100 will be listed in this page. Click on to edit signaling preference for FXS port or configure ACIM settings for FXO port.

Select "Loop Start" or "Kewl Start" for each FXS port. And then click on "Update" to save the change.



Figure 62: 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.

Tone Region

Advanced Settings

FXO Opermode

FXS Opermode

FXS TISS Override


Figure 63: FXO Ports ACIM Settings

Table 48: Internal Options/Hardware Config

Select country to set the default tones for dial tone, busy tone, ring tone and etc to be sent from the FXS port. 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

Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics. The default setting is "United


Select country to set the On Hook Speed, Ringer Impedance, Ringer

Threshold, Current Limiting, TIP/RING voltage adjustment, Minimum

Operational Loop Current, and AC Impedance as predefined for your country's analog line characteristics. The default setting is "United


Configure to enable or disable override Two-Wire Impedance Synthesis

(TISS). The default setting is No.


PCMA Override

Boost Ringer

Fast Ringer

Low Power

Ring Detect

FXS MWI Mode

If enabled, users can select the impedance value for Two-Wire

Impedance Synthesis (TISS) override. The default setting is 600

Ω.

Select the codec to be used for analog lines. North American users should choose PCMU. All other countries, unless already known, should 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 voltage (89V) for analog phones attached to the FXS port is required.

The default setting is "Normal".

Configure to increase the ringing speed to 25HZ. This option can be used with "Low Power" option. The default setting is "Normal".

Configure the peak voltage up to 50V during "Fast Ringer" operation.

This option is used with "Fast Ringer". The default setting is "Normal".

If set to "Full Wave", false ring detection will be prevented for lines where

Caller ID is sent before the first ring and proceeded by a polarity reversal, as in UK. The default setting is "Standard".

Configure the type of Message Waiting Indicator on FXS lines. The default setting is "FSK".



FSK: Frequency Shift Key Indicator



NEON: Light Neon Bulb Indicator.

INTERNAL OPTIONS/STUN MONITOR

Table 49: Internal Options/STUN Monitor

STUN Server

STUN Refresh

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.

Valid format:

[(hostname | IP-address) [':' port]

The default port number is 3478 if not specified.

Configure the number of seconds between STUN Refreshes. The default setting is 30 seconds.



IAX SETTINGS

The UCM6100 IAX global settings can be accessed via Web GUI->PBX->IAX Settings.

IAX SETTINGS/GENERAL

Bind Port

Bind Address

IAX1 Compatibility

No Checksums

Delay Reject

ADSI

Music On Hold Interpret

Music On Hold Suggest

Table 50: IAX Settings/General

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 calculated/checked on systems supporting this features. The default setting is "No".

If enabled, the IAX2 will delay the rejection of calls to avoid DOS. The default setting is "No".

Select to enable ADSI phone compatibility. The default setting is "No".

Specify which Music On Hold class this channel would like to listen to when 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.

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". Bandwidth

IAX SETTINGS/REGISTRATION

Table 51: IAX Settings/Registration

IAX Registration Options

Min Reg Expire

Max Reg Expire

IAX Thread Count

Configure the minimum period (in seconds) of registration. The default setting is 60.

Configure the maximum period (in seconds) of registration. The default setting is 3600.

Configure the number of IAX helper threads. The default setting is 10.



IAX Max Thread Count

Auto Kill

Authentication Debugging

Codec Priority

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 message is not received within 2000ms. Users could also specify number (in milliseconds) in addition to "yes" and "no". The default setting is "yes".

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.



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.

Configure ToS bit for preferred IP routing. Type of Service

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 52: IAX Settings/Static Defense

Call Token Optional

Max Call Numbers

Enter a single IP address or a range of IP addresses for which call token validation is not required.

For example:

11.11.11.11

11.11.11.11/22.22.22.22.

Configure the maximum number of calls allowed for a single IP address.



Max Unvalidated Call Numbers

Configure the maximum number of unvalidated calls for all IP addresses.

Call Number Limits

IP or IP Range

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.

For example:

11.11.11.11

11.11.11.11/22.22.22.22.



SIP SETTINGS

The UCM6100 SIP global settings can be accessed via Web GUI->PBX->SIP Settings.

SIP SETTINGS/GENERAL

Realm For Digest

Authentication

Bind UDP Port

Bind IP Address

Allow Guest Calls

Overlap Dialing

Allow Transfer

Enable DNS SRV Lookups on

Outbound Calls

MWI From

Table 53: SIP Settings/General

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".

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.

Select to enable overlap dialing support. The default setting is "No".

If set to "No", all transfers initiated by the endpoint in the UCM6100 will be disabled (unless enabled in peers or users). The default setting is

"Yes".

Select to enables DNS SRV lookups on outbound calls from the

UCM6100. The default setting is "Yes".

When sending MWI NOTIFY requests, this value will be used in the

"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

Domain

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 domains are accepted and sent to the default context or the context associated with the user/peer placing the call. Register to non-local domains will be automatically denied if a domain list is configured. Up to

10 domains can be added.



From Domain

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 54: SIP Settings/Misc

Outbound SIP Registrations

Register Timeout

Register Attempts

Configure the register retry timeout (in seconds). The default setting is

20.

Configure the number of registration attempts before the UCM6100 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

Reject Non-Matching INVITE

Configure the maximum bit rate (in kb/s) for video calls. The default setting is 384.

Select to enable video support in SIP calls. The default setting is "Yes".

If enabled, the UCM6100 will generate manager events when SIP UA performs events (e.g. Hold). The default setting is "No".

If enabled, when rejecting an incoming INVITE or REGISTER request, the UCM6100 will always reject with "401 Unauthorized" instead of notifying 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 55: SIP Settings/Session Timer

Session Timers

Select the session timer mode. The default setting is "Accept".

The options are:



Originate

Always request and run session timer.



Accept

Run session timer only when requested by other UA.



Session Expire

Min SE

Session Refresher



Refuse

Do not run session timer.

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

TCP Enable

TCP Bind Address

TLS Enable

TLS Bind Address

TLS Client Protocol

TLS Do Not Verify

TLS Self-Signed CA

Table 56: SIP Settings/TCP and TLS

Configure to allow incoming TCP connections with the UCM6100. The default setting is "No".

Configure the IP address for TCP server to bind to. 0.0.0.0 means binding to all interfaces. The port number is optional. If not specified,

5060 will be used.

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.

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

Select the TLS protocol for outbound client connections. The default setting is TLSv1.

If enabled, the TLS server's certificate won't be verified when acting as a client. The default setting is "Yes".

This is the CA certificate if the TLS server being connected to requires self-signed certificate, including server's public key. This file will be renames as "TLS.ca" automatically.



TLS Cert

TLS CA Cert

Note:

The size of the uploaded ca file must be under 2MB.

This is the Certificate file (*.pem format only) used for TLS connections.

It contains private key for client and signed certificate for the server. This file will be renamed as "TLS.pem" automatically.

Note:

The size of the uploaded certificate file must be under 2MB.

This file must be named with the CA subject name hash value. It contains CA's (Certificate Authority) public key, which is used to verify the accessed servers.

Note:

The size of the uploaded CA certificate file must be under 2MB.

Display a list of files under the CA Cert directory. TLS CA List

SIP SETTINGS/NAT

External IP Address

External Host

External Refresh

External TCP Port

External TLS Port

Local Network Address


Table 57: SIP Settings/NAT

Configure a static address and port (optional) that will be used in 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 except the host name will be looked up periodically based on the

"External Refresh" interval.

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 address will not be set correctly.

A sample configuration could be as follows:


NAT Mode

Allow RTP Reinvite

192.168.0.0/16

This is a global NAT setting that will affects all peers and users. The default setting is "Force rport".



No: Use rport if the remote side requires it.



Force rport: Force rport to always be on.



Yes: Force rport to be always on and perform comedia RTP handling.



Comedia: Use rport if the remote side requires it and performs comedia RTP handling.

Note:

"comedia RTP handling" refers to the technique of sending RTP to the port where the other endpoint's RTP packets come from. This can also be rephrased as "connection-oriented media".

If enabled, the UCM6100 will try to redirect the RTP media stream

(audio) to go directly from the caller to the callee. The default setting is

"No NAT".



Yes



No NAT: Allow media path redirection (Reinvite) but only when the peer is not be behind NAT. The RTP core can detect if the peer is behind NAT or not based on the IP address where the media comes from.



Update: Use UPDATE for media path redirection, instead of INVITE.

Note:

Some devices do not support this (especially if one of them is behind

NAT).

SIP SETTINGS/TOS

ToS For SIP

ToS For RTP Audio

ToS For RTP Video

Default Incoming/Outgoing

Registration Time


Table 58: SIP Settings/ToS

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.

Configure the default duration (in seconds) of incoming/outgoing registration. The default setting is 120.


Max Registration/Subscription

Time

Min Registration/Subscription

Time

Music On Hold Interpret

Music On Hold Suggest

Enable Relaxed DTMF

DTMF Mode

RTP Timeout

RTP Hold Timeout

Trust Remote Party ID

Send Remote Party ID

Generate In-Band Ringing

Configure the maximum duration (in seconds) of incoming registration and subscription allowed by the UCM6100. The default setting is 3600.

Configure the minimum duration (in seconds) of incoming registration and subscription allowed by the UCM6100. The default setting is 60.

Configure the Music On Hold class for the channel when being put on 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".

Configure the Music On Hold class to suggest to the peer channel when placing the peer on hold.

Select to enable relaxed DTMF handling. The default setting is "No".

Select DTMF mode to send DTMF. The default setting is RFC2833. If

"Info" is selected, SIP INFO message will be used. If "Inband" is selected, 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.

Note:

This setting doesn't apply to calls on hold.

When the call is on hold, if there is no RTP activity within the timeout (in seconds), the call will be terminated. This value of RTP Hold Timeout should be larger than RTP Timeout. The default setting is no timeout.

Configure whether the Remote-Party-ID should be trusted. The default setting is "No".

Configure whether the Remote-Party-ID should be sent or not. The default setting is "No".

Configure whether the 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.



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

Send Compact SIP Headers

Add "user=phone" to URI

Configure the user agent string for the UCM6100.

If enabled, compact SIP headers will be sent. The default setting is "No".

If enabled, "user=phone" will be added to URI that contains a valid phone number. The default setting is "No".



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 64: Status->PBX Status

TRUNKS

Users could see all the configured trunk status in this section.


Figure 65: Trunk Status


Table 59: Trunk Status

Status

Trunks

Type

Display trunk status.



Analog trunk status:

Available

Busy

Unavailable

Unknown Error



SIP Peer trunk status:

Unreachable: The hostname cannot be reached.

Unmonitored: QUALIFY feature is not turned on to be monitored.

Reachable: The hostname can be reached.



SIP Register trunk status:

Registered

Unrecognized Trunk

Display trunk name

Display trunk 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 to refresh the trunk status.



Click on [ + ] to expand the status detail table.



Click on [ - ] to hide the status detail table.

EXTENSIONS

Users could see all the configured extension status in this section.



Figure 66: Extension Status

Table 60: Extension Status

Status

Extension

Name/Label

Message

Type

Display extension number (including feature code). The color indicator has the following definitions.









Green: Free

Blue: Ringing

Yellow: In Use

Grey: Unavailable

Display the extension number.

Display name (callerID name) or label for the extension.

Display message status for the extension.

Example: 2/4/1

Description: There are 2 urgent messages, 4 messages in total and 1 message that has been already read.

Displays extension type.



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.





Click on one of the tabs accordingly.



Click on [ + ] to expand the status detail table.



Click on [ - ] to hide the status detail table.

to display the corresponding extensions

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 67: 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 61: 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 to refresh the call queue status.



Click on [ + ] to expand the call queue detail.



Click on [ - ] to hide the call queue detail.



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 68: 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 to refresh the conference room status.



Click on [ + ] to expand the conference room details.



Click on [ - ] to hide the conference room details.

INTERFACES STATUS

This section displays interface/port connection status on the UCM6100. The following example shows the interface status for UCM6116 with USB, SD card, LAN port and FXS1 connected.


Figure 69: UCM6116 Interfaces Status


Table 62: 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 hardware configuration page which can also be accessed via web GUI->PBX->Internal Options->Hardware Config.



Click on to refresh the interface status.



Click on [ + ] to expand the interface details.



Click on [ - ] to hide the interface details.

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 70: Parking Lot Status


Table 63: Parking Lot Status

Caller ID

Channel

Display the caller ID who parks the call.

Display channel for the call park.

Extension

Timeout

Display the parking lot number where the call is parked/retrieved.

Display timeout (in seconds) for the parked call. The status page will dynamically update this timer from 120 seconds (default) to 0. When the timer reaches 0, the 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 to refresh the parking lot status.



Click on [ + ] to expand the parking lot details.



Click on [ - ] to hide the parking details.

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 64: System Status->General

Status ->System Status -> General

Model

Part Number

Product model.

Product part number.



System Time

Up Time

Idle Time

Boot

Core

Base

Program

Recovery

Current system time. The current system time is also available on the upper right of each web page.

System up time since the last reboot.

System idle time since the last reboot.

Boot version.

Core version.

Base version.

Program version. This is the main software release version.

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 65: 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

Gateway

IP address.

Default gateway address.

Subnet Mask

DNS Server

Subnet mask address.

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.

Figure 71: 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 72: System Status->Resource Usage


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.

Click on to configure the parameters for each event.

1. Disk Usage

Figure 73: System Events->Alert Events Lists: Disk Usage



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. Modify Admin Password

Figure 74: System Events->Alert Events Lists: Modify Admin Password



Detect Cycle: The UCM6100 will initiate the admin password check based on this cycle. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.



3. Memory Usage

Figure 75: 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.

4. System Reboot

Figure 76: 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.

5. System Update

Figure 77: System Events->Alert Events Lists: System Update



Detect Cycle: The UCM6100 will check the system update based on this cycle. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.



6. System Crash

Figure 78: System Events->Alert Events Lists: System Crash



Detect Cycle: The UCM will detect the event at each cycle based on the specified time. Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.

Click on the switch to turn on/off the alert and Email notification for the event. Users could also select the checkbox for each event and then click on button "Alert On", "Alert Off", "Email Notification

On", "Email Notification Off" to control the alert and Email notification configuration.

ALERT LOG

Under Web GUI->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 79: 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.



Call Time. Format: 0:00:10.



Talk Time. Format: 0:00:10



Status. Format: NO ANSWER, BUSY, ANSWERED, or FAILED.



Options. Voice record playing/downloading/deleting.

Users could filter the call report by specifying the date range and criteria, depending on how the users would like to include the logs to the report. Then click on "View Report" button to display the generated report.

Figure 80: CDR Filter

Inbound calls

Outbound calls

Internal calls

External calls


Table 66: CDR Filter Criteria

Inbound calls are calls originated from a non-internal source (like a VoIP trunk) and sent to an internal extension.

Outbound calls are calls sent to a non-internal source (like a VoIP trunk) from an internal extension.

Internal calls are calls from one internal extension to another extension, which are not sent over a trunk.

External calls are calls sent from one trunk to another trunk, which are not sent to


Caller Number

Caller Name

From Date

To Date any internal extension.

Enter the caller number to be filtered in the CDR report.

Enter the caller name to be filtered in the CDR report.

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.

Specify "To" date and time to be filtered for the CDR report. Click on the field and the calendar will show for users to select the exact date and time.

The call report will display as the following figure shows.

Figure 81: 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 to play the recording file; click on to download the recording file in .wav format; click on to delete the recording file (the call record entry will not be deleted).

Figure 82: 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

"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 83: 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 84: 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 UCM6116, DAHDI/(1-16) are FXO ports, and DAHDI/(17-18) are FXS ports.

Sample 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:


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.

Trunk Type

Call Type

Time Range


Figure 87: CDR Statistics

Table 67: CDR Statistics Filter Criteria

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

The recording files recorded by "Auto Record" per extension/per trunk, or via feature code "Audio Mix

Record" are listed here. Users could click on to play the recording file; click on to download the recording file in .wav format; or 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.



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.

Upgrade Via


Figure 88: Network Upgrade

Table 68: Network Upgrade Configuration

Allow users to choose the firmware upgrade method: TFTP, HTTP or

HTTPS.


Firmware Server Path

Firmware File Prefix

Firmware File Suffix

Define the server path for the firmware server.

If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the UCM6100.

If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the UCM6100.

HTTP/HTTPS User Name

HTTP/HTTPS Password

The user name for the HTTP/HTTPS server.

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 the firmware file from your PC. The default firmware file name is ucm6100fw.bin

and select



Click on to start upgrading.

Figure 89: Local Upgrade



Figure 90: Upgrading Firmware Files



Wait until the upgrading process is successful and a window will be popped up in the Web GUI.

Figure 91: 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.



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.



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 92: 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.


Figure 93: Data Sync


Table 69: Data Sync Configuration

Enable Backup

Account

Password

Enable the auto backup function. The default setting is "No".

Enter the Account name on the SFTP backup server.

Enter the Password associate with the Account on the SFTP backup server.

Server Address

Backup Time

Enter the SFTP server address.

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 from the backup file.

to restore


Figure 94: 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.

Enable CDR Cleaner

CDR Clean Time

Clean Interval

Enable VR Cleaner


Figure 95: Cleaner

Table 70: Cleaner Configuration

Enable the CDR Cleaner function.

Enter 0-23 to specify the hour of the day to clean up CDR.

Enter 1-30 to specify the day of the month to clean up CDR.

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.

Enter 0-23 to specify the hour of the day to clean up Voice Records. VR Clean Time

Clean Interval Enter 1-30 to specify the day of the month to clean up Voice Records.

All the cleaner logs will be listed on the bottom of the page.

RESET AND REBOOT

Users could perform reset and reboot under Web GUI->Maintenance->Reset and Reboot.

To factory reset the device, select the mode type first. There are two different types for reset.



User Data: All the data including voicemail, recordings, IVR Prompt, Music on Hold, CDR and backup files will be cleared.



All: All the configurations and data will be reset to factory default.

Figure 96: 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.

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 97: 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 98: PING

TRACEROUTE

Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below.


Figure 99: Traceroute


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.



FCC Caution:

Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment.

This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions: (1)

This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.

Note: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

- Reorient or relocate the receiving antenna.

- Increase the separation between the equipment and receiver.

- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.

- Consult the dealer or an experienced radio/TV technician for help.

Regulatory Information

U.S. FCC Part 68 Statement

This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. The unit bears a label on the back which contains among other information a product identifier in the format US:

GNIIS00BUCM6104/US: GNIIS00BUCM6116. If requested, this number must be provided to the telephone company.

This equipment uses the following standard jack types for network connection: RJ11C.

This equipment contains an FCC compliant modular jack. It is designed to be connected to the telephone network or premises wiring using compatible modular plugs and cabling which comply with the requirements of FCC Part 68 rules.

The Ringer Equivalence Number, or REN, is used to determine the number of devices which may be connected to the telephone line. An excessive REN may cause the equipment to not ring in response to an incoming call. In most areas, the sum of the RENs of all equipment on a line should not exceed five (5.0).



In the unlikely event that this equipment causes harm to the telephone network, the telephone company can temporarily disconnect your service. The telephone company will try to warn you in advance of any such disconnection, but if advance notice isn't practical, it may disconnect the service first and notify you as soon as possible afterwards. In the event such a disconnection is deemed necessary, you will be advised of your right to file a complaint with the FCC.

From time to time, the telephone company may make changes in its facilities, equipment, or operations which could affect the operation of this equipment. If this occurs, the telephone company is required to provide you with advance notice so you can make the modifications necessary to obtain uninterrupted service.

There are no user serviceable components within this equipment. See Warranty flyer for repair or warranty information.

It shall be unlawful for any person within the United States to use a computer or other electronic device to send any message via a telephone facsimile unless such message clearly contains, in a margin at the top or bottom of each transmitted page or on the first page of the transmission, the date and time it is sent and an identification of the business, other entity, or individual sending the message and the telephone number of the sending machine or of such business, other entity, or individual. The telephone number provided may not be a 900 number or any other number for which charges exceed local or long distance transmission charges. Telephone facsimile machines manufactured on and after December 20, 1992, must clearly mark such identifying information on each transmitted message. Facsimile modem boards manufactured on and after December 13, 1995, must comply with the requirements of this section.

This equipment cannot be used on public coin phone service provided by the telephone company.

Connection to Party Line Service is subject to state tariffs. Contact your state public utility commission, public service commission, or corporation commission for more information.



Grandstream Networks UCM6104 User manual

Grandstream Networks, Inc.

UCM6100 Series IP PBX

User Manual

UCM6100 Series IP PBX User Manual

Index

CHANGE LOG

FIRMWARE VERSION 1.0.5.19

FIRMWARE VERSION 1.0.5.14

FIRMWARE VERSION 1.0.4.7

FIRMWARE VERSION 1.0.3.13

FIRMWARE VERSION 1.0.2.21

FIRMWARE VERSION 1.0.1.22

WELCOME

PRODUCT OVERVIEW

FEATURE HIGHTLIGHTS

TECHNICAL SPECIFICATIONS

Interfaces

Voice/Video Capabilities

Signaling and Control

Security

Physical

Additional Features

INSTALLATION

EQUIPMENT PACKAGING

CONNECT YOUR UCM6100

SAFETY COMPLIANCES

WARRANTY

GETTING STARTED

USE THE LCD MENU

USE THE LED INDICATORS

USE THE WEB GUI

MAKE YOUR FIRST CALL

SYSTEM SETTINGS

NETWORK SETTINGS

FIREWALL

CHANGE PASSWORD

LDAP SERVER

HTTP SERVER

EMAIL SETTINGS

TIME SETTINGS

NTP SERVER

PROVISIONING

OVERVIEW

AUTO PROVISIONING

MANUAL PROVISIONING

EXTENSIONS

CREATE NEW USER

BATCH ADD EXTENSIONS

EDIT EXTENSION

TRUNKS

ANALOG TRUNKS

VOIP TRUNKS

CALL ROUTES

OUTBOUND ROUTES

INBOUND ROUTES

CONFERENCE BRIDGE

IVR

CONFIGURE IVR

CREATE IVR PROMPT

LANGUAGE SETTINGS FOR VOICE PROMPT

DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE

CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE

VOICEMAIL

CONFIGURE VOICEMAIL

VOICEMAIL EMAIL SETTINGS

CONFIGURE VOICEMAIL GROUP

RING GROUP

CONFIGURE RING GROUP

PAGING AND INTERCOM GROUP

CONFIGURE PAGING/INTERCOM GROUP

CALL QUEUE

CONFIGURE CALL QUEUE

EXTENSION GROUPS

CONFIGURE EXTENSION GROUPS

USE EXTENSION GROUPS

PICKUP GROUPS

CONFIGURE PICKUP GROUPS

MUSIC ON HOLD

FAX/T.38