experiencing the ucm6100 series ip pbx

Grandstream Networks, Inc.

UCM6100 Series IP PBX

User Manual

UCM6100 SERIES IP PBX USER MANUAL

UCM6100 Series IP PBX User Manual

Index

CHANGE LOG ........................................................................................... 14

FIRMWARE VERSION 1.0.7.10 .......................................................................................................... 14

FIRMWARE VERSION 1.0.6.11 .......................................................................................................... 14

FIRMWARE VERSION 1.0.6.10 .......................................................................................................... 14

FIRMWARE VERSION 1.0.5.19 .......................................................................................................... 15

FIRMWARE VERSION 1.0.5.14 .......................................................................................................... 15

FIRMWARE VERSION 1.0.4.7 ............................................................................................................ 16

FIRMWARE VERSION 1.0.3.13 .......................................................................................................... 16

FIRMWARE VERSION 1.0.2.21 .......................................................................................................... 17

FIRMWARE VERSION 1.0.1.22 .......................................................................................................... 18

WELCOME ................................................................................................ 19

PRODUCT OVERVIEW ............................................................................. 20

FEATURE HIGHTLIGHTS ................................................................................................................... 20

TECHNICAL SPECIFICATIONS ......................................................................................................... 20

INSTALLATION ......................................................................................... 23

EQUIPMENT PACKAGING ................................................................................................................. 23

CONNECT YOUR UCM6100 .............................................................................................................. 23

CONNECT THE UCM6102 ........................................................................................................... 23

CONNECT THE UCM6104 ........................................................................................................... 25

CONNECT THE UCM6108 ........................................................................................................... 25

CONNECT THE UCM6116 ........................................................................................................... 26

SAFETY COMPLIANCES ................................................................................................................... 27

WARRANTY ........................................................................................................................................ 27

GETTING STARTED ................................................................................. 28

USE THE LCD MENU ......................................................................................................................... 28

USE THE LED INDICATORS .............................................................................................................. 30

USE THE WEB GUI ............................................................................................................................ 31

ACCESS WEB GUI ...................................................................................................................... 31

WEB GUI CONFIGURATIONS ..................................................................................................... 32

WEB GUI LANGUAGES ............................................................................................................... 33

SAVE AND APPLY CHANGES ..................................................................................................... 34

MAKE YOUR FIRST CALL .................................................................................................................. 34

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 1 of 239

SYSTEM SETTINGS ................................................................................. 35

NETWORK SETTINGS ....................................................................................................................... 35

BASIC SETTINGS ........................................................................................................................ 35

802.1X .......................................................................................................................................... 39

STATIC ROUTES ......................................................................................................................... 39

PORT FORWORDING (UCM6102 ONLY) ................................................................................... 41

FIREWALL ........................................................................................................................................... 41

STATIC DEFENSE ....................................................................................................................... 41

DYNAMIC DEFENSE ................................................................................................................... 44

FAIL2BAN ..................................................................................................................................... 44

CHANGE PASSWORD ....................................................................................................................... 45

LDAP SERVER .................................................................................................................................... 46

LDAP SERVER CONFIGURATIONS ........................................................................................... 46

LDAP PHONEBOOK .................................................................................................................... 48

LDAP CLIENT CONFIGURATIONS ............................................................................................. 49

HTTP SERVER .................................................................................................................................... 51

EMAIL SETTINGS ............................................................................................................................... 51

TIME SETTINGS ................................................................................................................................. 52

AUTO TIME UPDATING ............................................................................................................... 52

SET TIME MANUALLY ................................................................................................................. 54

OFFICE TIME ............................................................................................................................... 55

HOLIDAY ...................................................................................................................................... 56

NTP SERVER ...................................................................................................................................... 58

PROVISIONING ......................................................................................... 59

OVERVIEW ......................................................................................................................................... 59

CONFIGURATION ARCHITECTURE FOR END POINT DEVICE ...................................................... 59

AUTO PROVISIONING SETTINGS .................................................................................................... 60

DISCOVERY ........................................................................................................................................ 63

GLOBAL CONFIGURATION ............................................................................................................... 64

GLOBAL POLICY ......................................................................................................................... 64

GLOBAL TEMPLATES ................................................................................................................. 70

MODEL CONFIGURATION ................................................................................................................. 72

DEVICE CONFIGURATION ................................................................................................................ 74

CREATE NEW DEVICE ................................................................................................................ 74

MANAGE DEVICES ..................................................................................................................... 75

SAMPLE APPLICATION ...................................................................................................................... 79

EXTENSIONS ............................................................................................ 83

CREATE NEW USER .......................................................................................................................... 83

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CREATE NEW SIP EXTENSION ................................................................................................. 83

CREATE NEW IAX EXTENSION ................................................................................................. 87

CREATE NEW FXS EXTENSION ................................................................................................ 91

BATCH ADD EXTENSIONS ................................................................................................................ 96

BATCH ADD SIP EXTENSIONS .................................................................................................. 96

BATCH ADD IAX EXTENSIONS .................................................................................................. 98

EDIT EXTENSION ............................................................................................................................. 101

EXPORT EXTENSIONS .................................................................................................................... 102

IMPORT EXTENSIONS..................................................................................................................... 102

TRUNKS .................................................................................................. 104

ANALOG TRUNKS ............................................................................................................................ 104

ANALOG TRUNK CONFIGURATION ........................................................................................ 104

PSTN DETECTION .................................................................................................................... 107

VOIP TRUNKS .................................................................................................................................. 110

Direct Outward Dialing (DOD) .................................................................................................... 119

SLA STATION .......................................................................................... 121

CREATE/EDIT SLA STATION ........................................................................................................... 121

SAMPLE CONFIGURATION ............................................................................................................. 122

CALL ROUTES ........................................................................................ 125

OUTBOUND ROUTES ...................................................................................................................... 125

INBOUND ROUTES .......................................................................................................................... 127

INBOUND RULE CONFIGURATIONS ....................................................................................... 128

BLACKLIST CONFIGURATIONS ............................................................................................... 130

CONFERENCE BRIDGE ......................................................................... 132

CONFERENCE BRIDGE CONFIGURATIONS .......................................................................... 132

JOIN A CONFERENCE CALL .................................................................................................... 134

INVITE OTHER PARTIES TO JOIN CONFERENCE ................................................................. 134

DURING THE CONFERENCE ................................................................................................... 135

RECORD CONFERENCE .......................................................................................................... 136

IVR ........................................................................................................... 138

CONFIGURE IVR .............................................................................................................................. 138

CREATE CUSTOM PROMPT ........................................................................................................... 140

RECORD NEW CUSTOM PROMPT .......................................................................................... 140

UPLOAD CUSTOM PROMPT .................................................................................................... 141

LANGUAGE SETTINGS FOR VOICE PROMPT .................................... 142

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DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE ............................................................... 142

CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE .............................................................. 144

VOICEMAIL ............................................................................................. 145

CONFIGURE VOICEMAIL ................................................................................................................ 145

VOICEMAIL EMAIL SETTINGS ........................................................................................................ 146

CONFIGURE VOICEMAIL GROUP .................................................................................................. 147

RING GROUP .......................................................................................... 149

CONFIGURE RING GROUP ............................................................................................................. 149

PAGING AND INTERCOM GROUP ........................................................ 152

CONFIGURE PAGING/INTERCOM GROUP .................................................................................... 152

CALL QUEUE .......................................................................................... 154

CONFIGURE CALL QUEUE ............................................................................................................. 154

EXTENSION GROUPS ............................................................................ 158

CONFIGURE EXTENSION GROUPS .............................................................................................. 158

USE EXTENSION GROUPS ............................................................................................................. 159

PICKUP GROUPS ................................................................................... 160

CONFIGURE PICKUP GROUPS ...................................................................................................... 160

MUSIC ON HOLD .................................................................................... 161

FAX/T.38 .................................................................................................. 163

CONFIGURE FAX/T.38 ..................................................................................................................... 163

SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE ............................................... 164

SAMPLE CONFIGURATION FOR FAX-TO-EMAIL .......................................................................... 166

FOLLOW ME ........................................................................................... 168

ONE-KEY DIAL ....................................................................................... 170

DISA ........................................................................................................ 171

BLF AND EVENT LIST ............................................................................ 173

BLF .................................................................................................................................................... 173

EVENT LIST ...................................................................................................................................... 173

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DIAL BY NAME ....................................................................................... 176

DIAL BY NAME CONFIGURATION................................................................................................... 176

CALL FEATURES ................................................................................... 180

FEATURE CODES ............................................................................................................................ 180

CALL RECORDING ........................................................................................................................... 183

CALL PARK ....................................................................................................................................... 184

PARK A CALL ............................................................................................................................. 184

RETRIEVE THE PARKED CALL ................................................................................................ 184

INTERNAL OPTIONS .............................................................................. 185

INTERNAL OPTIONS/GENERAL ..................................................................................................... 185

INTERNAL OPTIONS/JITTER BUFFER ........................................................................................... 187

INTERNAL OPTIONS/RTP SETTINGS ............................................................................................ 187

INTERNAL OPTIONS/PORTS CONFIG ........................................................................................... 188

INTERNAL OPTIONS/STUN MONITOR ........................................................................................... 190

INTERNAL OPTIONS/PAYLOAD ...................................................................................................... 190

IAX SETTINGS ........................................................................................ 191

IAX SETTINGS/GENERAL................................................................................................................ 191

IAX SETTINGS/REGISTRATION ...................................................................................................... 191

IAX SETTINGS/STATIC DEFENSE .................................................................................................. 192

SIP SETTINGS ........................................................................................ 194

SIP SETTINGS/GENERAL ................................................................................................................ 194

SIP SETTINGS/MISC ........................................................................................................................ 195

SIP SETTINGS/SESSION TIMER ..................................................................................................... 195

SIP SETTINGS/TCP and TLS ........................................................................................................... 196

SIP SETTINGS/NAT .......................................................................................................................... 197

SIP SETTINGS/TOS.......................................................................................................................... 198

STATUS AND REPORTING .................................................................... 200

PBX STATUS ..................................................................................................................................... 200

TRUNKS ..................................................................................................................................... 200

EXTENSIONS ............................................................................................................................. 201

QUEUES ..................................................................................................................................... 202

CONFERENCE ROOMS ............................................................................................................ 203

INTERFACES STATUS .............................................................................................................. 204

PARKING LOT ............................................................................................................................ 205

ACTIVITY CALLS .............................................................................................................................. 206

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SYSTEM STATUS ............................................................................................................................. 207

GENERAL ................................................................................................................................... 207

NETWORK ................................................................................................................................. 208

STORAGE USAGE ..................................................................................................................... 208

RESOURCE USAGE .................................................................................................................. 209

SYSTEM EVENTS ............................................................................................................................ 210

ALERT EVENTS LIST ................................................................................................................ 210

ALERT LOG ................................................................................................................................ 212

ALERT CONTACT ...................................................................................................................... 212

CDR ................................................................................................................................................... 212

DOWNLOADED CDR FILE ........................................................................................................ 214

STATISTICS ................................................................................................................................ 216

RECORDING FILES ................................................................................................................... 217

CDR API CONFIGURATION FILES ........................................................................................... 218

UPGRADING AND MAINTENANCE ....................................................... 224

UPGRADING ..................................................................................................................................... 224

UPGRADING VIA NETWORK .................................................................................................... 224

UPGRADING VIA LOCAL UPLOAD ........................................................................................... 225

NO LOCAL FIRMWARE SERVERS ........................................................................................... 227

BACKUP ............................................................................................................................................ 227

LOCAL BACKUP ........................................................................................................................ 227

DATA SYNC ................................................................................................................................ 228

RESTORE CONFIGURATION FROM BACKUP FILE ............................................................... 229

CLEANER .......................................................................................................................................... 230

RESET AND REBOOT ...................................................................................................................... 232

SYSLOG ............................................................................................................................................ 232

TROUBLESHOOTING ...................................................................................................................... 233

ETHERNET CAPTURE .............................................................................................................. 233

IP PING ....................................................................................................................................... 233

TRACEROUTE ........................................................................................................................... 234

ANALOG RECORD TRACE ....................................................................................................... 234

SERVICE CHECK ...................................................................................................................... 235

EXPERIENCING THE UCM6100 SERIES IP PBX .................................. 237

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 6 of 239

Table of Tables

UCM6100 Series IP PBX User Manual

Table 1: Technical Specifications ................................................................................................................ 20

Table 2: UCM6102/UCM6104 Equipment Packaging ................................................................................. 23

Table 3: UCM6108/UCM6116 Equipment Packaging ................................................................................. 23

Table 4: LCD Menu Options ........................................................................................................................ 29

Table 5: UCM6102/UCM6104 LED INDICATORS ...................................................................................... 30

Table 6: UCM6108/UCM6116 LED INDICATORS ...................................................................................... 31

Table 7: UCM6102 Network Settings->Basic Settings ................................................................................ 35

Table 8: UCM6104 Network Settings->Basic Settings ................................................................................ 37

Table 9: UCM6108/UCM6116 Network Settings->Basic Settings ............................................................... 39

Table 10: UCM6100 Network Settings->802.1X ......................................................................................... 39

Table 11: UCM6100 Network Settings->Static Routes ................................................................................ 40

Table 12: UCM6102 Network Settings->Port Forwarding ........................................................................... 41

Table 13: UCM6100 Firewall->Static Defense->Current Service ................................................................ 41

Table 14: Typical Firewall Settings .............................................................................................................. 42

Table 15: Firewall Rule Settings .................................................................................................................. 43

Table 16: UCM6102 Firewall Dynamic Defense ......................................................................................... 44

Table 17: Fail2Ban Settings ........................................................................................................................ 45

Table 18: HTTP Server Settings .................................................................................................................. 51

Table 19: Email Settings .............................................................................................................................. 51

Table 20: Time Auto Updating ..................................................................................................................... 53

Table 21: Create New Office Time .............................................................................................................. 55

Table 22: Create New Holiday .................................................................................................................... 57

Table 23: Auto Provision Settings ............................................................................................................... 62

Table 24: Global Policy Parameters: Localization ....................................................................................... 65

Table 25: Global Policy Parameters: Contact List ....................................................................................... 66

Table 26: Global Policy Parameters: Maintenance ..................................................................................... 67

Table 27: Global Policy Parameters: Network Settings .............................................................................. 69

Table 28: Global Policy Parameters: Customization ................................................................................... 69

Table 29: Create New Template .................................................................................................................. 71

Table 30: Create New Model Template ....................................................................................................... 72

Table 31: SIP Extension Configuration Parameters .................................................................................... 83

Table 32: IAX Extension Configuration Parameters .................................................................................... 87

Table 33: FXS Extension Configuration Parameters .................................................................................. 91

Table 34: Batch Add SIP Extension Parameters ......................................................................................... 96

Table 35: Batch Add IAX Extension Parameters ......................................................................................... 99

Table 36: Analog Trunk Configuration Parameters ................................................................................... 104

Table 37: PSTN Detection for Analog Trunk ............................................................................................. 109

Table 38: SIP Trunk Configuration Parameters ......................................................................................... 111

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Table 39: IAX Trunk Configuration Parameters ......................................................................................... 116

Table 40: SLA Station Configuration Parameters ...................................................................................... 121

Table 41: Outbound Route Configuration Parameters .............................................................................. 125

Table 42: Inbound Rule Configuration Parameters ................................................................................... 128

Table 43: Conference Bridge Configuration Parameters .......................................................................... 132

Table 44: Conference Caller IVR Menu .................................................................................................... 136

Table 45: IVR Configuration Parameters .................................................................................................. 138

Table 46: Voicemail Settings ..................................................................................................................... 145

Table 47: Voicemail Email Settings ........................................................................................................... 147

Table 48: Voicemail Group Settings .......................................................................................................... 148

Table 49: Ring Group Parameters ............................................................................................................ 149

Table 50: Paging/Intercom Group Configuration Parameters ................................................................... 152

Table 51: Call Queue Configuration Parameters ...................................................................................... 154

Table 52: FAX/T.38 Settings ...................................................................................................................... 163

Table 53: Follow Me Settings .................................................................................................................... 169

Table 54: Follow Me Options ..................................................................................................................... 169

Table 55: DISA Settings ............................................................................................................................ 171

Table 56: Event List Settings ..................................................................................................................... 174

Table 57: UCM6100 Feature Codes ......................................................................................................... 180

Table 58: Internal Options/General ........................................................................................................... 185

Table 59: Internal Options/Jitter Buffer ...................................................................................................... 187

Table 60: Internal Options/RTP Settings ................................................................................................... 187

Table 61: Internal Options/Ports Config .................................................................................................... 188

Table 62: Internal Options/STUN Monitor ................................................................................................. 190

Table 63: Internal Options/Payload ........................................................................................................... 190

Table 64: IAX Settings/General ................................................................................................................. 191

Table 65: IAX Settings/Registration .......................................................................................................... 191

Table 66: IAX Settings/Static Defense ...................................................................................................... 192

Table 67: SIP Settings/General ................................................................................................................. 194

Table 68: SIP Settings/Misc ...................................................................................................................... 195

Table 69: SIP Settings/Session Timer ....................................................................................................... 195

Table 70: SIP Settings/TCP and TLS ........................................................................................................ 196

Table 71: SIP Settings/NAT ....................................................................................................................... 197

Table 72: SIP Settings/ToS ........................................................................................................................ 198

Table 73: Trunk Status ............................................................................................................................... 200

Table 74: Extension Status ........................................................................................................................ 202

Table 75: Agent Status .............................................................................................................................. 203

Table 76: Interface Status Indicators ......................................................................................................... 204

Table 77: Parking Lot Status ..................................................................................................................... 205

Table 78: System Status->General ........................................................................................................... 207

Table 79: System Status->Network ........................................................................................................... 208

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 8 of 239

Table 80: CDR Filter Criteria ..................................................................................................................... 213

Table 81: CDR Statistics Filter Criteria ...................................................................................................... 217

Table 82: CDR API Configuration Files ..................................................................................................... 218

Table 83: CDR API URI Parameters ......................................................................................................... 219

Table 84: Network Upgrade Configuration ................................................................................................ 224

Table 85: Data Sync Configuration ........................................................................................................... 229

Table 86: Cleaner Configuration ............................................................................................................... 231

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 9 of 239

Table of Figures

UCM6100 Series IP PBX User Manual

Figure 1: UCM6102 Front View ................................................................................................................... 24

Figure 2: UCM6102 Back View ................................................................................................................... 24

Figure 3: UCM6104 Front View ................................................................................................................... 25

Figure 4: UCM6104 Back View ................................................................................................................... 25

Figure 5: UCM6108 Front View ................................................................................................................... 26

Figure 6: UCM6108 Back View ................................................................................................................... 26

Figure 7: UCM6116 Front View ................................................................................................................... 26

Figure 8: UCM6116 Back View ................................................................................................................... 26

Figure 9: UCM6116 Web GUI Login Page .................................................................................................. 31

Figure 10: UCM6100 Web GUI Language .................................................................................................. 33

Figure 11: Create New Firewall Rule .......................................................................................................... 43

Figure 12: LDAP Server Configurations ...................................................................................................... 47

Figure 13: Default LDAP Phonebook DN .................................................................................................... 47

Figure 14: Default LDAP Phonebook Attributes .......................................................................................... 48

Figure 15: Add LDAP Phonebook ............................................................................................................... 48

Figure 16: Edit LDAP Phonebook ............................................................................................................... 49

Figure 17: GXP2200 LDAP Phonebook Configuration ............................................................................... 50

Figure 18: UCM6100 Email Settings ........................................................................................................... 52

Figure 19: Set Time Manually ..................................................................................................................... 54

Figure 20: Create New Office Time ............................................................................................................. 55

Figure 21: Settings->Time Settings->Office Time ....................................................................................... 56

Figure 22: Create New Holiday ................................................................................................................... 57

Figure 23: Settings->Time Settings->Holiday ............................................................................................. 58

Figure 24: Zero Config Configuration Architecture for End Point Device ................................................... 60

Figure 25: UCM6100 Zero Config ............................................................................................................... 61

Figure 26: Auto Provision Settings .............................................................................................................. 62

Figure 27: Auto Discover ............................................................................................................................. 64

Figure 28: Discovered Devices ................................................................................................................... 64

Figure 29: Discovered Devices ................................................................................................................... 65

Figure 30: Edit Global Template .................................................................................................................. 71

Figure 31: Edit Model Template .................................................................................................................. 73

Figure 32: Create New Device .................................................................................................................... 74

Figure 33: Manage Devices ........................................................................................................................ 75

Figure 34: Edit Device ................................................................................................................................. 76

Figure 35: Edit Customize Device Settings ................................................................................................. 77

Figure 36: Add P Value in Customize Device Settings ............................................................................... 78

Figure 37: Device List in Zero Config .......................................................................................................... 78

Figure 38: Zero Config Sample - Global Policy ........................................................................................... 79

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 10 of 239

Figure 39: Zero Config Sample - Device Preview 1 .................................................................................... 80

Figure 40: Zero Config Sample - Device Preview 2 .................................................................................... 81

Figure 41: Zero Config Sample - Device Preview 3 .................................................................................... 82

Figure 42: Export Extensions .................................................................................................................... 102

Figure 43: Import Extensions .................................................................................................................... 102

Figure 44: UCM6100 FXO Tone Settings ................................................................................................. 107

Figure 45: UCM6100 PSTN Detection ...................................................................................................... 108

Figure 46: UCM6100 PSTN Detection: Auto Detect ................................................................................. 108

Figure 47: UCM6100 PSTN Detection: Semi-Auto Detect ....................................................................... 109

Figure 48: DOD extension selection ......................................................................................................... 119

Figure 49: Edit DOD .................................................................................................................................. 120

Figure 50: SLA Station .............................................................................................................................. 121

Figure 51: Enable SLA Mode for Analog Trunk ......................................................................................... 123

Figure 52: Analog Trunk with SLA Mode Enabled .................................................................................... 123

Figure 53: SLA Example - SLA Station ..................................................................................................... 123

Figure 54: SLA Example - MPK Configuration .......................................................................................... 124

Figure 55: Blacklist Configuration Parameters .......................................................................................... 130

Figure 56: Blacklist csv File ....................................................................................................................... 131

Figure 57: Conference Invitation From Web GUI ...................................................................................... 134

Figure 58: Conference Recording ............................................................................................................. 137

Figure 59: Click on Prompt to Create IVR Prompt .................................................................................... 140

Figure 60: Record New Custom Prompt ................................................................................................... 140

Figure 61: Upload Custom Prompt ........................................................................................................... 141

Figure 62: Language Settings For Voice Prompt ...................................................................................... 143

Figure 63: Voice Prompt Package List ...................................................................................................... 143

Figure 64: New Voice Prompt Language Added ....................................................................................... 144

Figure 65: Voicemail Email Settings ......................................................................................................... 146

Figure 66: Voicemail Group ....................................................................................................................... 147

Figure 67: Ring Group ............................................................................................................................... 149

Figure 68: Ring Group Configuration ........................................................................................................ 151

Figure 69: Paging/Intercom Group ............................................................................................................ 152

Figure 70: Page/Intercom Group Settings ................................................................................................ 153

Figure 71: Call Queue ............................................................................................................................... 154

Figure 72: Agent Login Settings ................................................................................................................ 157

Figure 73: Edit Extension Group ............................................................................................................... 158

Figure 74: Select Extension Group in Outbound Route ............................................................................ 159

Figure 75: Edit Pickup Group .................................................................................................................... 160

Figure 76: Music On Hold Default Class ................................................................................................... 161

Figure 77: Configure Analog Trunk without Fax Detection ....................................................................... 164

Figure 78: Configure Extension for Fax Machine ..................................................................................... 165

Figure 79: Configure Inbound Rule for Fax ............................................................................................... 165

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Figure 80: Create Fax Extension .............................................................................................................. 166

Figure 81: Inbound Route to Fax Extension ............................................................................................. 166

Figure 82: Create Follow Me ..................................................................................................................... 168

Figure 83: Edit Follow Me ......................................................................................................................... 168

Figure 84: Configure One-Key Dial ........................................................................................................... 170

Figure 85: Create New DISA..................................................................................................................... 171

Figure 86: Create New Event List ............................................................................................................. 174

Figure 87: Create Dial By Name Group .................................................................................................... 176

Figure 88: Dial By Name Group In IVR Key Pressing Events .................................................................. 177

Figure 89: Dial By Name Group In Inbound Rule ..................................................................................... 178

Figure 90: Configure Extension First Name and Last Name .................................................................... 179

Figure 91: Download Recording File From CDR Page ............................................................................. 183

Figure 92: FXS Ports Signaling Preference .............................................................................................. 188

Figure 93: FXO Ports ACIM Settings ........................................................................................................ 188

Figure 94: Status->PBX Status ................................................................................................................. 200

Figure 95: Trunk Status ............................................................................................................................. 200

Figure 96: Extension Status ...................................................................................................................... 201

Figure 97: Queue Status ........................................................................................................................... 203

Figure 98: Conference Room Status ......................................................................................................... 204

Figure 99: UCM6116 Interfaces Status ..................................................................................................... 204

Figure 100: Parking Lot Status .................................................................................................................. 205

Figure 101: Status->PBX Status->Activity Calls: Calling .......................................................................... 206

Figure 102: Status->PBX Status->Activity Calls ........................................................................................ 206

Figure 103: System Status->Storage Usage ............................................................................................. 209

Figure 104: System Status->Resource Usage .......................................................................................... 209

Figure 105: System Events->Alert Events Lists: Disk Usage ................................................................... 210

Figure 106: System Events->Alert Events Lists: Memory Usage ............................................................. 211

Figure 107: System Events->Alert Events Lists: System Reboot ............................................................. 211

Figure 108: System Events->Alert Events Lists: System Crash ............................................................... 211

Figure 109: System Events->Alert Log ..................................................................................................... 212

Figure 110: CDR Filter .............................................................................................................................. 213

Figure 111: Call Report.............................................................................................................................. 214

Figure 112: Call Report Entry with Audio Recording File .......................................................................... 214

Figure 113: Downloaded CDR File Sample - Call To Shows "s" ............................................................... 215

Figure 114: Downloaded CDR File Sample - Source Channel and Dest Channel 1 ................................ 215

Figure 115: Downloaded CDR File Sample - Source Channel and Dest Channel 2 ................................ 215

Figure 116: Downloaded CDR File Sample - Source Channel and Dest Channel 3 ................................ 216

Figure 117: CDR Statistics ........................................................................................................................ 216

Figure 118: CDR->Recording Files ........................................................................................................... 217

Figure 119: Network Upgrade ................................................................................................................... 224

Figure 120: Local Upgrade ........................................................................................................................ 225

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Figure 121: Upgrading Firmware Files ...................................................................................................... 226

Figure 122: Reboot UCM6100 .................................................................................................................. 226

Figure 123: Local Backup ......................................................................................................................... 228

Figure 124: Data Sync .............................................................................................................................. 229

Figure 125: Restore UCM6100 from Backup File ..................................................................................... 230

Figure 126: Cleaner .................................................................................................................................. 231

Figure 127: Reset and Reboot .................................................................................................................. 232

Figure 128: Ethernet Capture ................................................................................................................... 233

Figure 129: PING ...................................................................................................................................... 234

Figure 130: Traceroute .............................................................................................................................. 234

Figure 131: Troubleshooting Analog Trunks ............................................................................................. 235

Figure 132: Service Check ........................................................................................................................ 236

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 13 of 239

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 1.0.7.10

Added office time and holiday setting. [OFFICE TIME] [HOLIDAY]

Added new architecture for auto provisioning via Zero Config. [PROVISIONING]

Added Alert-info configuration for distinctive ringing on inbound route. [Table 42: Inbound Rule

Configuration Parameters]

Added SLA support to monitor FXO trunk. [SLA STATION]

Added Follow Me support. [FOLLOW ME]

Added One-Key Dial support. [ONE-KEY DIAL]

Added FXO Dial Delay option for analog trunk. [Table 36: Analog Trunk Configuration

Parameters]

Added Music On Hold selection per extension. [EXTENSIONS]

Added custom prompt support for ring group announcement. [Table 49: Ring Group Parameters]

Added custom prompt support for paging/intercom group announcement. [Table 50:

Paging/Intercom Group Configuration Parameters]

Added options to delete voicemail and Fax files via cleaner. [CLEANER]

Added DTMF configuration per trunk. [VOIP TRUNKS]

Added call queue Music On Hold customization. [Table 51: Call Queue Configuration Parameters]

Added call queue destination support. [Table 51: Call Queue Configuration Parameters]

FIRMWARE VERSION 1.0.6.11

Added service check options under web UI->Maintenance->Troubleshooting->Service Check page to regularly check Asterisk service.

FIRMWARE VERSION 1.0.6.10

Added static routes function. [STATIC ROUTES]

Added option to provision end devices’ date format, time format and time zone in zero config.

Added option to disable extension/trunk.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 14 of 239

Added TEL URL support for extension/trunk.

Added option to dial trunk password per extensions.

Added export extension and import extension function. [EXPORT EXTENSIONS] [IMPORT

EXTENSIONS]

Added option "Need Registration" for SIP register trunk.

Added option "The Maximum Number of Call Line" for trunk.

Added Dial By Name. [DIAL BY NAME]

Added voicemail password and Email address for voicemail group extension.

Added auto record support for ring group and call queue.

Added VFax file display, download and delete interface in web UI.

Changed web page name from "Hardware Config" to "Ports Config".

Added payload configuration for audio/video codecs. [INTERNAL OPTIONS/PAYLOAD]

Added activity calls status on web UI status page. [ACTIVITY CALLS]

Added CDR API support. [CDR API CONFIGURATION FILES]

Added more alert events support such as Register SIP Failed, Register SIP Trunk Failed, Restore

Config, User Login Success, User Login Failed, SIP Internal Call Failure and etc. [ALERT

EVENTS LIST]

FIRMWARE VERSION 1.0.5.19

Added built-in data migration tool to support upgrading from 1.0.4.7 to 1.0.5.19 without factory reset.

Added "Direct Dial Voicemail Prefix" feature code back. [Table 57: UCM6100 Feature Codes]

Changed valid range for option "Current Disconnect Threshold". [Table 36: Analog Trunk

Configuration Parameters]

FIRMWARE VERSION 1.0.5.14

New backend data structure and web UI performance improvement. 1.0.5.14 is not compatible with previous firmware versions. Once upgraded to 1.0.5.14, the device needs to be FULLY

RESET and RE-CONFIGURED MANUALLY.

Added traditional Chinese language for web UI. [WEB GUI LANGUAGES]

Updated LDAP configuration example. [LDAP SERVER]

Added "Enable Filter Source Caller ID" and "Custom Dynamic Route" options for outbound route

settings. [Table 41: 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 49: Ring Group Parameters]

Added "Extension Groups" section in web UI. [EXTENSION GROUPS]

Added "Pickup Groups" section in web UI. [PICKUP GROUPS]

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 15 of 239

Added BLF function description. [BLF AND EVENT LIST]

Updated default extension range. [Table 58: Internal Options/General]

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

FIRMWARE VERSION 1.0.4.7

Asterisk updated to version 1.8.23.1.

Added DID routing support for incoming calls. [Table 38: SIP Trunk Configuration Parameters]

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

Added GXP one-button Voicemail access. [Table 31: SIP Extension Configuration Parameters]

Added option "Skip voicemail password verification" on extension edit page. [Table 31: SIP

Extension Configuration Parameters]

Added Hot-Desking Support. [Table 31: 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 61:

Internal Options/Ports Config]

Added more modes for FXS Two-Wire Impedance Synthesis

Added LDAP Sync manual trigger function and synced date displaying. [VOIP TRUNKS]

Improved LDAP Sync function, added retrying, file verifying and progress displaying function

Added option "Pick Extension Period" on auto-provision settings page of Zero Config

Added multiple extension assignment support on device edit page of Zero Config

Added "Reset All Extensions" button at the Zero Config page to recycle all assigned extensions.

[AUTO PROVISIONING]

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

[Figure 108: 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 38: SIP Trunk Configuration Parameters]

FIRMWARE VERSION 1.0.3.13

Added Fail2Ban support for SIP authentication. [FAIL2BAN]

Added voice prompt "Language" selection and "Auto Record" option for extension. [Table 31: SIP

Extension Configuration Parameters] [Table 34: Batch Add SIP Extension Parameters]

Added "Auto Record" option for trunk.

Added "Digit Timeout" option and voice prompt "Language" selection for IVR. [Table 45: IVR

Configuration Parameters]

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

[Table 57: UCM6100 Feature Codes]

Added "Enforce Strong Passwords" option. [Table 58: Internal Options/General]

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 16 of 239

Added FXS MWI Mode. [Table 61: Internal Options/Ports Config]

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

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

FIRMWARE VERSION 1.0.2.21

Added weight information. [Table 1: Technical Specifications]

Added NTP server support. [NTP SERVER]

Added Czech language for web GUI display. [WEB GUI LANGUAGES]

Added VLAN support. "Layer 2 QoS 802.1Q/VLAN tag" and "Layer 2 QoS 802.1p priority value"

options for network port settings are added. [NETWORK SETTINGS]

Updated LDAP client configurations information.

Added sample Email settings. [EMAIL SETTINGS]

Added manual time settings. [TIME SETTINGS]

Added "Enable Pick Extension" and "Extension Segment" options for auto provisioning settings.

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

(NOTIFY)" in zero-config feature.

Added pickup group feature. [Table 57: 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 38: SIP Trunk Configuration

Parameters]

Added LDAP sync options for peer SIP trunk. [Table 38: SIP Trunk Configuration Parameters]

Changed the default setting of outbound route "Privilege Level" from "Internal" to "International" to avoid potential misconfiguration and security risk.

Added DISA and Fax to IVR key press event options. [Table 45: IVR Configuration Parameters]

Added "Min Message Time" option in voicemail settings. [Table 46: Voicemail Settings]

Added Fax setting samples. [FAX/T.38]

Added DISA support for inbound route and IVR.

Added Event List support to monitor local extensions and remote extensions. [BLF AND EVENT LIST]

Added feature code *0 for "Disconnect". [Table 57: UCM6100 Feature Codes]

Added feature code *8 for "Pickup Extension" in pickup group feature. [Table 57: UCM6100 Feature

Codes]

Added "Record Prompt" and "Custom Name of Pickup Group" options in internal options. [Table 58:

Internal Options/General]

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

misconfiguration. [Table 64: IAX Settings/General]

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

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 17 of 239

FIRMWARE VERSION 1.0.1.22

This is the initial version.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 18 of 239

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.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 19 of 239

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

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 20 of 239

LCD Display

Reset Switch

Voice/Video Capabilities

128x32 graphic LCD with DOWN and OK button

Yes

Voice-over-Packet

Capabilities

Voice and Fax Codecs

Video Codecs

QoS

Signaling and Control

DTMF Methods

Provisioning Protocol and

Plug-and-Play

Network Protocols

Disconnect Methods

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

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

UCM6104: Unit weight 0.51kg, Package weight 0.94kg

UCM6108: Unit weight 2.23kg, Package weight 3.09kg

UCM6116: Unit weight 2.27kg, Package weight 3.14kg

UCM6102/6104: Wall mount and Desktop

UCM6108/6116: Rack mount and Desktop

Additional Features

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 21 of 239

Multi-language Support

Caller ID

Polarity Reversal/ Wink

Call Center

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

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

UCM6102/6104: Up to 3 password-protected conference bridges

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)

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 22 of 239

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

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 23 of 239

Figure 1: UCM6102 Front View

Figure 2: UCM6102 Back View

To set up the UCM6102, follow the steps below:



Connect one end of an RJ-45 Ethernet cable into the WAN port of the UCM6102.



Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.



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.



Wait for the UCM6102 to boot up. The LCD in the front will show the device hardware information when the boot process is done.



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.



(Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and

Fax) to the FXS ports.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 24 of 239

CONNECT THE UCM6104

Figure 3: UCM6104 Front View

Figure 4: UCM6104 Back View

To set up the UCM6104, follow the steps below:



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



Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.



Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6104. Insert the main plug of the power adapter into a surge-protected power outlet.



Wait for the UCM6104 to boot up. The LCD in the front will show the device hardware information when the boot process is done.



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.



(Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and

Fax) to the FXS ports.

CONNECT THE UCM6108

To set up the UCM6108, follow the steps below:

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 25 of 239



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



Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.



Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6108. Insert the main plug of the power adapter into a surge-protected power outlet.



Wait for the UCM6108 to boot up. The LCD in the front will show the device hardware information when the boot process is done.



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.



(Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and

Fax) to the FXS ports.

Figure 5: UCM6108 Front View

CONNECT THE UCM6116

Figure 6: UCM6108 Back View

Figure 7: UCM6116 Front View

Firmware Version 1.0.7.10

Figure 8: UCM6116 Back View

UCM6100 Series IP PBX User Manual Page 26 of 239

To set up the UCM6116, follow the steps below:



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



Connect the other end of the Ethernet cable into the uplink port of an Ethernet switch/hub.



Connect the 12V DC power adapter into the 12V DC power jack on the back of the UCM6116. Insert the main plug of the power adapter into a surge-protected power outlet.



Wait for the UCM6116 to boot up. The LCD in the front will show the device hardware information when the boot process is done.



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.



(Optional) Connect PSTN lines from the wall jack to the FXO ports; connect analog lines (phone and

Fax) to the FXS ports.

SAFETY COMPLIANCES

The UCM6100 series IP PBX complies with FCC/CE and various safety standards. The UCM6100 power adapter is compliant with the UL standard. Use the universal power adapter provided with the UCM6100 package only. The manufacturer’s warranty does not cover damages to the device caused by unsupported power adapters.

WARRANTY

If the UCM6100 series IP PBX was purchased from a reseller, please contact the company where the device was purchased for replacement, repair or refund. If the device was purchased directly from

Grandstream, contact our Technical Support Team for a RMA (Return Materials Authorization) number before the product is returned. Grandstream reserves the right to remedy warranty policy without prior notification.

Warning:

Use the power adapter provided with the UCM6100 series IP PBX. Do not use a different power adapter as this may damage the device. This type of damage is not covered under warranty.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 27 of 239

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.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 28 of 239

The following table shows the LCD menu options.

Table 4: LCD Menu Options

View Events

Critical Events

Other Events

Device Info

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

Network Info

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

Network Menu

Factory Menu

For UCM6104/UCM6108/UCM6116:

LAN Mode: Select LAN mode as DHCP, Static IP or PPPoE

Static Route Reset: Click to reset the static route setting

For UCM6102:

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

Static Route Reset: Click to reset the static route setting

Reboot

Factory Reset

LCD Test Patterns

Press "OK" to start. Then press "Down" button to test different LCD

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 29 of 239

Web Info

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

Fan Mode

Select "Auto" or "On".

LED Test Patterns

Select "All On" "All Off" or "Blinking" and check LED status.

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

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 30 of 239

LED

NETWORK

Table 6: UCM6108/UCM6116 LED INDICATORS

LED Status

Solid: Connected

OFF: Not Connected

ACT

USB

SD

Phone (FXS)

Line (FXO)

Solid: Connected

Flashing: Data Transferring

OFF: Not Connected

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.

Firmware Version 1.0.7.10

Figure 9: UCM6116 Web GUI Login Page

UCM6100 Series IP PBX User Manual Page 31 of 239

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 ad ministrator’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.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 32 of 239

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.

Firmware Version 1.0.7.10

Figure 10: UCM6100 Web GUI Language

UCM6100 Series IP PBX User Manual Page 33 of 239

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.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 34 of 239

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

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

"Gateway IP" for this interface.

Preferred DNS Server Enter the preferred DNS server address.

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

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

IP Address

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

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

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

255.255.0.0.

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

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

0.0.0.0.

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.

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

IP Address

Subnet Mask

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

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

255.255.0.0.

Gateway IP

DNS Server 1

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

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

0.0.0.0.

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DNS Server 2

User Name

Password

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.

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

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

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

IP Address

Subnet Mask

Gateway IP

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.

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

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

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

255.255.0.0.

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

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

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 LAN port. The default value is 0.

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

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default interface in option "Default Interface". Users will need assign LAN 1 or LAN 2 as the default interface in option "Default Interface" and configure

"Gateway IP" for this interface.

Preferred DNS Server Enter the preferred DNS server address.

LAN (when Method is set to "Switch")

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

IP Address

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

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

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

255.255.0.0.

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

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

0.0.0.0.

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 LAN port. The default value is 0.

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

If "Dual" is selected as "Method", users will need assign the default interface to be LAN 1 or LAN 2. The default interface is LAN 2.

IP Method

IP Address

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

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

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

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

255.255.0.0.

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

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

0.0.0.0.

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 LAN port. The default value is 0.

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

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

Preferred DNS Server Enter the preferred DNS server address.

IP Method

IP Address

Subnet Mask

Gateway IP

DNS Server 1

DNS Server 2

User Name

Password

Layer 2 QoS

802.1Q/VLAN Tag

Layer 2 QoS 802.1p

Priority Value

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

Enter the IP address for static IP settings.

Enter the subnet mask address for static IP settings.

Enter the gateway IP address for static IP settings.

Enter the DNS server 1 address for static IP settings.

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 LAN port. The default value is 0.

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

802.1X

The UCM6100 provides users 802.1X settings for LAN port and WAN port (WAN port: UCM6102 only).

Table 10: UCM6100 Network Settings->802.1X

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.

STATIC ROUTES

The UCM6100 provides users static routing capability that allows the device to use manually configured routes, rather than information only from dynamic routing or gateway configured in the UCM6100 web

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GUI->Network Settings->Basic Settings to forward traffic. It can be used to define a route when no other routes are available or necessary, or used in complementary with existing routing on the UCM6100 as a failover backup, and etc.

Click on in the table below.

to create a new static route. The configuration parameters are listed

Once added, users can select to edit the static route.

Select to delete the static route.

Table 11: UCM6100 Network Settings->Static Routes

Destination

Netmask

Gateway

Configure the destination IP address or the destination IP subnet for the

UCM6100 to reach using the static route.

Example:

IP address - 192.168.66.4

IP subnet - 192.168.66.0

Configure the subnet mask for the above destination address. If left blank, the default value is 255.255.255.255.

Example:

255.255.255.0

Configure the gateway address so that the UCM6100 can reach the destination via this gateway. Gateway address is optional.

Example:

192.168.40.5

Interface

Specify the network interface on the UCM6100 to reach the destination using the static route.

For UCM6102, LAN interface is eth0; WAN interface is eth1.

For UCM6104, LAN1 interface is eth0; WAN interface is eth1.

For UCM6108/UCM6116, only LAN interface is available.

Static routes configuration can be reset from LCD menu->Network Menu.

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PORT FORWORDING (UCM6102 ONLY)

The UCM6102 network interface supports router functions which provides users the ability to do port forwarding. If the UCM6102 LAN mode is set to "Route" under web GUI->Settings->Network Settings-

>Basic Settings page, port forwarding is available for configuration.

The port forwarding configuration is under web GUI->Settings->Network Settings->Port Forwarding page. Please see related settings in the table below.

Table 12: 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 13: UCM6100 Firewall->Static Defense->Current Service

Port

7777

Process

Asterisk

Type

tcp/IPv4

Protocol or Service

SIP

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389

22

80

8089

69

9090

6060

5060

4569

5353

Slapd

Dropbear

Lighthttpd

Lighthttpd

Opentftpd

Asterisk zero_config

Asterisk

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

LDAP

SSH

HTTP

HTTPS

TFTP

SIP

UCM6100 zero_config service

SIP

SIP

UCM6100 zero_config service

37435 Syslogd udp/IPv4 Syslog

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

Table 14: Typical Firewall Settings

Ping Defense

Enable

SYN-Flood Defense

Enable

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

Enable to prevent SYN Flood denial-of-service attack to the device. The default setting is disabled. To enable or disable it, click on the check box for the LAN or WAN (UCM6102 only) interface.

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.

Right next to "Create New Rule" button, there is a checkbox for option "Reject Rules". If it’s checked, all the rules will be rejected except the firewall rules listed below. In the firewall rules, only when there is a rule that meets all the following requirements, the option "Reject Rules" will be allowed to check:

Action: "Accept"

Type "In"

Destination port is set to the system login port (e.g., by default 8089)

Protocol is not UDP

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Figure 11: Create New Firewall Rule

Table 15: Firewall Rule Settings

Rule Name

Action

Type

Service

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

Select the service type.

FTP

SSH

Telnet

TFTP

HTTP

LDAP

Custom

If "Custom" is selected, users will need specify Source (IP and port),

Destination (IP and port) and Protocol (TCP, UDP or Both) for the service.

Please note if the source or the destination field is left blank, it will be used as "Anywhere".

Save the change, 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,

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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 16: 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 and 59 when dynamic defense is turned on. The default setting is

59.

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

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

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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 17: 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

Protocol

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.

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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.

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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

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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.

Firmware Version 1.0.7.10

Figure 15: Add LDAP Phonebook

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Configure the "Phonebook Prefix" first. The "Phonebook DN" will be automatically filled in. For example, if configuring "Phonebook Prefix" as "people", the "Phonebook DN" will be filled with

"ou=people,dc=pbx,dc=com".

Once added, users can select 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

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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

.

Firmware Version 1.0.7.10

Figure 17: GXP2200 LDAP Phonebook Configuration

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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 18: HTTP Server Settings

Redirect From Port 80

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".

Protocol Type

Port

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

Specify port number to access the HTTP server. The default port number is 8089.

Once the change is saved, the web page will be redirected to the login page using the new URL. Enter the username and password to login again.

EMAIL SETTINGS

The Email application on the UCM6100 can be used to send out alert event Emails, Fax (Fax-To-Email),

Voicemail (Voicemail-To-Email) and etc. The configuration parameters can be accessed via Web GUI-

>Settings->Email Settings.

Table 19: Email Settings

TLS Enable

Type

Domain

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".

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Server

Username

Password

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".

Specify the display name in the FROM header in the Email. Display Name

Sender

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

AUTO TIME UPDATING

The current system time on the UCM6100 is displayed on the upper right of the web page. It can also be found under Web GUI->Status->System Status->General.

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To configure the UCM6100 to update time automatically, go to Web GUI->Settings->Time Settings-

>Time Auto Updating.

Note:

The configurations under Web GUI->Settings->Time Settings->Time Auto Updating page require reboot to take effect. Please consider configuring auto time updating related changes when setting up the

UCM6100 for the first time to avoid service interrupt after installation and deployment in production.

Table 20: 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

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

M4.1.0,M11.1.0

The 1st number indicates Month: 1,2,3.., 12 (for Jan, Feb, .., Dec).

The 2nd number indicates the nth iteration of the weekday: (1st Sunday,

3rd Tuesday…). Normally 1, 2, 3, 4 are used. If 5 is used, it means the last iteration of the weekday.

The 3rd number indicates weekday: 0,1,2,..,6 ( for Sun, Mon, Tues, ...

,Sat).

Therefore, this example is the DST which starts from the First Sunday of

April to the 1st Sunday of November.

SET TIME MANUALLY

To manually set the time on the 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

Note:

Manually setup time will take effect immediately after saving and applying change in the web UI. If users would like to reboot the UCM6100 and keep the manually setup time setting, please make sure "Remote

NTP Server", "Enable DHCP Option 2" and "Enable DHCP Option 42" options under Web GUI->Settings-

>Time Settings->Time Auto Updating page are unchecked or set to empty. Otherwise, time auto updating settings in this page will take effect after reboot.

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OFFICE TIME

On the UCM6100, the system administrator can define "office time", which can be used to configure time condition for extension call forwarding schedule and inbound rule schedule. To configure office time, go to

Web GUI->Settings->Time Settings->Office Time. Click on "Create New Office Time" to create an office time.

Figure 20: Create New Office Time

Start Time

End Time

Week

Show Advanced Options

Month

Day

Table 21: Create New Office Time

Configure the start time for office hour.

Configure the end time for office hour

Select the work days in one week.

Check this options to show advanced options. Once selected, please specify "Month" and "Day" below.

Select the months for office time.

Select the work days in one month.

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Select "Start Time", "End Time" and the day for the "Week" for the office time. The system administrator can also define month and day of the month as advanced options. Once done, click on "Save" and then

"Apply Change" for the office time to take effect. The office time will be listed in the web page as the figure shows below.

Figure 21: Settings->Time Settings->Office Time

Click on to edit the office time.

Click on to delete the office time.

Click on "Delete Selected Office Times" to delete multiple selected office times at once.

HOLIDAY

On the UCM6100, the system administrator can define "holiday", which can be used to configure time condition for extension call forwarding schedule and inbound rule schedule. To configure holiday, go to

Web GUI->Settings->Time Settings->Holiday. Click on "Create New Holiday" to create holiday time.

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Figure 22: Create New Holiday

Table 22: Create New Holiday

Name

Holiday Memo

Month

Day

Specify the holiday name to identify this holiday.

Create a note for the holiday.

Select the month for the holiday.

Select the day for the holiday.

Show Advanced Options

Check this option to show advanced options. If selected, please specify the days as holiday in one week below.

Select the days as holiday in one week. Week

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

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Figure 23: Settings->Time Settings->Holiday

Click on to edit the holiday.

Click on to delete the holiday.

Click on "Delete Selected Holidays" to delete multiple selected holidays at once.

NTP SERVER

The UCM6100 can be used as a NTP server for the NTP clients to synchronize their time with. To configure the UCM6100 as the NTP server, set "Enable NTP server" to "Yes" under web GUI->Settings-

>Time Settings->NTP Server. On the client side, point the NTP server address to the UCM6100 IP address or host name to use the UCM6100 as the NTP server.

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PROVISIONING

OVERVIEW

Grandstream SIP Devices can be configured via Web interface as well as via configuration file through

TFTP/HTTP/HTTPS download. All Grandstream SIP devices support a proprietary binary format configuration file and XML format configuration file. The UCM6100 provides a Plug and Play mechanism to auto-provision the Grandstream SIP devices in a zero configuration manner by generating XML config file and having the phone to download it within LAN area. This allows users to finish the installation with ease and start using the SIP devices in a managed way.

To provision a phone, three steps are involved, i.e., discovery, configuration and provisioning. This section explains how Zero Config works on the UCM6100. The settings for this feature can be accessed via Web

GUI->PBX->Zero Config.

CONFIGURATION ARCHITECTURE FOR END POINT DEVICE

Started from firmware version 1.0.7.10, the end point device configuration in zero config is divided into the following three layers with priority from the lowest to the highest:

Global

This is the lowest layer. Users can configure the most basic options that could apply to all Grandstream

SIP devices during provisioning via Zero config.

Model

In this layer, users can define model-specific options for the configuration template.

Device

This is the highest layer. Users can configure device-specific options for the configuration for individual device here.

Each layer also has its own structure in different levels. Please see figure below. The details for each layer are explained in sections

[GLOBAL CONFIGURATION]

,

[MODEL CONFIGURATION]

and

[DEVICE

CONFIGURATION]

.

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Figure 24: Zero Config Configuration Architecture for End Point Device

The configuration options in model layer and device layer have all the option in global layers already, i.e., the options in global layer is a subset of the options in model layer and device layer. If an option is set in all three layers with different values, the highest layer value will override the value in lower layer. For example, if the user selects English for Language setting in Global Policy and Spanish for Language setting in Default Model Template, the language setting on the device to be provisioned will use Spanish as model layer has higher priority than global layer. To sum up, configurations in higher layer will

always override the configurations for the same options/fields in the lower layer when presented

at the same time.

After understanding the zero config configuration architecture, users could configure the available options for end point devices to be provisioned by the UCM6100 by going through the three layers. This configuration architecture allows users to set up and manage the Grandstream end point devices in the same LAN area in a centralized way.

AUTO PROVISIONING SETTINGS

By default, the Zero Config feature is enabled on the UCM6100 for auto provisioning. Three methods of auto provisioning are used.

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Figure 25: UCM6100 Zero Config

SIP SUBSCRIBE

When the phone boots up, it sends out SUBSCRIBE to a multicast IP address in the LAN. The

UCM6100 discovers it and then sends a NOTIFY with the XML config file URL in the message body.

The phone will then use the path to download the config file generated in the UCM6100 and take the new configuration.

DHCP OPTION 66

This method should be used on the UCM6102 because only the UCM6102 has WAN and LAN port with LAN port supporting the router function. When the phone restarts (by default DHCP Option 66 is turned on), it will send out a DHCP DISCOVER request. The UCM6102 receives it and returns DHCP

OFFER with the config server path URL in Option 66, for example, https://192.168.2.1:8089/zccgi/.

The phone will then use the path to download the config file generated in the UCM6100.

mDNS

When the phone boots up, it sends out mDNS query to get the TFTP server address. The UCM6100 will respond with its own address. The phone will then send TFTP request to download the XML config file from the UCM6100.

To start the auto provisioning process, under Web GUI->PBX->Zero Config->Auto Provision Settings, fill in the auto provision information.

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Figure 26: Auto Provision Settings

Table 23: Auto Provision Settings

Enable Zero Config

Enable Automatic

Configuration Assignment

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

By default, this is disabled. If disabled, when SIP device boots up, the

UCM6100 will not send the SIP device the URL to download the config file and therefore the SIP device will not be automatically provisioned by the UCM6100.

Note:

When disabled, SIP devices can still be provisioned by manually sending

NOTIFY from the UCM6100 which will include the XML config file URL for the SIP device to download.

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. Zero Config Extension

Segment range can be defined in web UI->PBX->Internal Options-

>General page->Extension Preference section: "Auto Provision

Extensions".

If enabled, the extension list will be sent out to the device after receiving

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Pick Extension Segment

Pick Extension Period (hour): the device's request. This feature is for the GXP series phones that support selecting extension to be provisioned via phone's LCD. The default setting is disabled.

Click on the link "Pick Extension Segment" to specify the extension list to be sent to the device. The default range is 4000 to 4999. Pick Extension

Segment range can be defined in web UI->PBX->Internal Options-

>General page->Extension Preference section: "Pick Extensions".

Specify the number of minutes to allow the phones being provisioned to pick extensions.

Please make sure an extension is manually assigned to the phone or "Automatically Assign Extension" is enabled during provisioning. After the configuration on the UCM6100 web GUI, click on "Save" and "Apply

Changes". Once the phone boots up and picks up the config file from the UCM6100, it will take the configuration right away.

DISCOVERY

Users could manually discover the device by specifying the IP address or scanning the entire LAN network. Three methods are supported to scan the devices.

PING

ARP

SIP Message (NOTIFY)

Click on "Auto Discover", fill in the "Scan Method" and "Scan IP". The IP address segment will be automatically filled in based on the network mask detected on the UCM6100. If users need scan the entire network segment, enter 255 (for example, 192.168.40.255) instead of a specific IP address. Then click on "Save" to start discovering the devices within the same network. To successfully discover the devices, "Zero Config" needs to be enabled on the UCM6100 web GUI->PBX->Zero Config->Auto

Provisioning Settings.

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Figure 27: 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 28: Discovered Devices

GLOBAL CONFIGURATION

GLOBAL POLICY

Global configuration will apply to all the connected Grandstream SIP end point devices in the same LAN with the UCM6100 no matter what the Grandstream device model it is. It is divided into two levels:

Web UI->PBX->Zero Config->Global Policy

Web UI->PBX->Zero Config->Global Templates.

Global Templates configuration has higher priority to Global Policy configuration.

Global Policy can be accessed in web GUI->PBX->Zero Config->Global Policy page. On the top of the configuration table, users can select category in the "Options" dropdown list to quickly navigate to the category. The categories are:

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Localization: configure display language, data and time.

Contact List: configure LDAP and XML phonebook download.

Maintenance: configure upgrading, web access, Telnet/SSH access and syslog.

Network Settings: configure IP address, QoS and STUN settings.

Customization: customize LCD screen wallpaper for the supported models.

Figure 29: Discovered Devices

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

The following tables list the Global Policy configuration parameters for the SIP end device.

Table 24: Global Policy Parameters: Localization

Language settings

Language

Date and Time

Date Format

Select the LCD display language on the SIP end device.

Time Format

Configure the date display format on the SIP end device

’s LCD.

Configure the time display in 12-hour or 24-hour format on the SIP end device’s LCD.

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NTP Server

Time Zone

LDAP Phonebook

Source

Address

Port

Base DN

User Name

Password

Number Filter

Name Filter

Version

Name Attribute

Number Attribute

Display Name

Firmware Version 1.0.7.10

Configure the URL or IP address of the NTP server. The SIP end device may obtain the date and time from the server.

Configure the time zone used on the SIP end device.

Table 25: Global Policy Parameters: Contact List

Select "Manual" or "PBX" as the LDAP configuration source.

If "Manual" is selected, the LDAP configuration below will be applied to the SIP end device.

If "PBX" is selected, the LDAP configuration built-in from UCM6510 web UI->Settings->LDAP Server will be applied.

Configure the IP address or DNS name of the LDAP server.

Configure the LDAP server port. The default value is 389.

This is the location in the directory where the search is requested to begin. Example:

 dc=grandstream, dc=com

 ou=Boston, dc=grandstream, dc=com

Configure the bind "Username" for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

Configure the bind "Password" for querying LDAP servers. The field can be left blank if the LDAP server allows anonymous binds.

Configure the filter used for number lookups. Please refer to user manual for more details.

Configure the filter used for name lookups. Please refer to user manual for more details.

Select the protocol version for the phone to send the bind requests. The default value is 3.

Specify the "name" attributes of each record which are returned in the

LDAP search result. Example:

 gn

 cn sn description

Specify the "number" attributes of each record which are returned in the

LDAP search result. Example:

 telephoneNumber

 telephoneNumber Mobile

Configure the entry information to be shown on phone's LCD. Up to 3 fields can be displayed. Example:

%cn %sn %telephoneNumber

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Max Hits

Search Timeout

Sort Results

Incoming Calls

Outgoing Calls

Lookup Display Name

Specify the maximum number of results to be returned by the LDAP server. Valid range is 1 to 3000. The default value is 50.

Specify the interval (in seconds) for the server to process the request and client waits for server to return. Valid range is 0 to 180. The default value is 30.

Specify whether the searching result is sorted or not. The default setting is No.

Configure to enable LDAP number searching when receiving calls. The default setting is No.

Configure to enable LDAP number searching when making calls. The default setting is No.

Configures the display name when LDAP looks up the name for incoming call or outgoing call. It must be a subset of the LDAP Name Attributes.

XML Phonebook

Remove manually-edited entries on download

Phonebook XML Server

Phonebook Download Interval

Select the source of the phonebook XML server.

Disable

Disable phonebook XML downloading.

Manual

Once selected, users need specify downloading protocol HTTP,

HTTPS or TFTP and the server path to download the phonebook

XML file. The server path could be IP address or URL, with up to 256 characters.

Local UCM Server

Once selected, click on the Server Path field to upload the phonebook XML file. Please note: after uploading the phonebook

XML file to the server, the original file name will be used as the directory name and the file will be renamed as phonebook.xml under that directory.

Configure the phonebook download interval (in Minute). If set to 0, automatic download will be disabled. Valid range is 5 to 720.

If set to "Yes", when XML phonebook is downloaded, the entries added manually will be automatically removed.

Table 26: Global Policy Parameters: Maintenance

Upgrade and Provision

Upgrade via

Server Path

Select upgrade via TFTP, HTTP or HTTPS.

Configure the firmware upgrading server path.

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File Prefix

File Posfix

Allow DHCP Option 43/66

Automatic Upgrade

Firmware Upgrade Rule

Web Access

Admin Password

End-User Password

Web Access Mode

Web Server Port

Security

Disable Telnet/SSH

Syslog

Syslog Server

Syslog Level

Send SIP Log

Firmware Version 1.0.7.10

If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the phone.

If configured, only the configuration file with the matching encrypted postfix will be downloaded and flashed into the phone.

If DHCP option 43 or 66 is enabled on the LAN side, the TFTP server can be redirected.

If enabled, UCM6100 will automatically upgrade if a new firmware is detected. Users can select automatic upgrading by day, by week or by minute.

By week

Once selected, specify the day of the week to check HTTP/TFTP server for firmware upgrades or configuration files changes.

By day

Once selected, specify the hour of the day to check the HTTP/TFTP server for firmware upgrades or configuration files changes.

By minute

Once selected, specify the interval X that the SIP end device will request for new firmware every X minutes.

Specify how firmware upgrading and provisioning request to be sent.

Configure the administrator password for admin level login.

Configure the end-user password for the end user level login.

Select HTTP or HTTPS as the web access protocol.

Configure the port for web access. The valid range is 1 to 65535.

Enable Telnet/SSH access for the SIP end device. If the SIP end device supports Telnet access, this option controls the Telnet access of the device; if the SIP end device supports SSH access, this option controls the SSH access of the device.

Configure the URL/IP address for the syslog server.

Select the level of logging for syslog.

Configure whether the SIP log will be included in the syslog message.

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Table 27: Global Policy Parameters: Network Settings

Basic Settings

IP Address

Configure how the SIP end device shall obtain the IP address. DHCP or

PPPoE can be selected.

DHCP

Once selected, users can specify the Host Name (option 12) of the

SIP end device as DHCP client, and Vendor Class ID (option 60) used by the client and server to exchange vendor class ID information.

PPPoE

Once selected, users need specify the Account ID, Password and

Service Name for PPPoE.

Advanced Setting

Layer 3 QoS

Layer 2 QoS Tag

Layer 2 QoS Priority Value

STUN Server

Keep Alive Interval

Define the Layer 3 QoS parameter. This value is used for IP Precedence,

Diff-Serv or MPLS. Valid range is 0-63.

Assign the VLAN Tag of the Layer 2 QoS packets. Valid range is 0 -4095.

Assign the priority value of the Layer 2 QoS packets. Valid range is 0-7.

Configure the IP address or Domain name of the STUN server. Only nonsymmetric NAT routers work with STUN.

Specify how often the phone will send a blank UDP packet to the SIP server in order to keep the "ping hole" on the NAT router to open. Valid range is 10-160.

Table 28: Global Policy Parameters: Customization

Wallpaper

Screen Resolution 1024 x 600

Screen Resolution 800 x 400

Check this option if the SIP end device shall use 1024 x 600 resolution for the LCD screen wallpaper.

Source

Configure the location where wallpapers are stored.

File

If "URL" is selected as source, specify the URL of the wallpaper file.

If "Local UCM Server" is selected as source, click to upload wallpaper file to the UCM6100.

Check this option if the SIP end device shall use 800 x 400 resolution for the LCD screen wallpaper.

Source

Configure the location where wallpapers are stored.

File

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Screen Resolution 480 x 272

Screen Resolution 320 x 240

If "URL" is selected as source, specify the URL of the wallpaper file.

If "Local UCM Server" is selected as source, click to upload wallpaper file to the UCM6100.

Check this option if the SIP end device shall use 480 x 272 resolution for the LCD screen wallpaper.

Source

Configure the location where wallpapers are stored.

File

If "URL" is selected as source, specify the URL of the wallpaper file.

If "Local UCM Server" is selected as source, click to upload wallpaper file to the UCM6100.

Check this option if the SIP end device supports 320 x 240 resolution for the LCD screen wallpaper.

Source

Configure the location where wallpapers are stored.

File

If "URL" is selected as source, specify the URL of the wallpaper file.

If "Local UCM Server" is selected as source, click to upload wallpaper file to the UCM6100.

GLOBAL TEMPLATES

Global Templates can be accessed in web GUI->PBX->Zero Config->Global Templates. Users can create multiple global templates with different sets of configurations and save the templates. Later on, when the user configures the device in Edit Device dialog->Advanced Settings, the user can select to use one of the global template for the device. Please refer to section

[MANAGE DEVICES]

for more details on using the global templates.

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

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

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Template Name

Description

Enabled

Table 29: Create New Template

Create a name to identify this global template.

Provide a description for the global template. This is optional.

Check this option to enable the global template.

Click on to edit the global template.

The window for editing global template is shown in the following figure.

In the “Options” field, after entering the option name key word, the options containing the key word will be listed. Users could then select the options to be modified and click on “Add Option” to add it into the global template.

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

Figure 30: Edit Global Template

Click on “Save” to save this global template.

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The created global templates will show in the web UI->PBX->Zero Config->Global Templates page.

Users can click on to delete the global template or click on

“Delete Selected Templates” to delete multiple selected templates at once.

Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected templates.

MODEL CONFIGURATION

Model layer configuration allows users to apply model-specific configurations to different devices. Users could create/edit/delete a model template by accessing web GUI, page PBX->Zero Config->Model

Templates. If multiple model templates are created and enabled, when the user configures the device in

Edit Device dialog->Advanced Settings, the user can select to use one of the model template for the device. Please refer to section

[MANAGE DEVICES]

for more details on using the model template.

For each created model template, users can assign it as default model template. If assigned as default model template, the values in this model template will be applied to all the devices of this model. There is always only one default model template that can be assigned at one time on the UCM6100.

The selected model template and the default model template will both take effect when generating the config file for the device. However the model template has higher priority to default model template when it comes to the same setting option/field. If the same option/field has different value configured in the default model template and the selected model template, the value for this option/field in the selected model template will override the value in default model template.

Click on

“Create New Template” to add a model template.

Table 30: Create New Model Template

Model

Template Name

Description

Default Model Template

Enabled

Select a model to apply this template. The supported Grandstream models are listed in the dropdown list for selection.

Create a name for the model template.

Enter a description for the model template. This is optional.

Select to assign this model template as the default model template. The value of the option in default model template will be overridden if other selected model template has a different value for the same option.

Check this option to enable the model template.

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Click on to edit the model template.

The editing window for model template is shown in the following figure.

In the “Options” field, enter the option name key word, the option that contains the key word will be listed. User could then select the option and click on “Add Option” to add it into the model template.

Once added, the option will be shown in the list below. On the left side of each option, users can click on to remove this option from the model template. On the right side of each option, users can click on to reset the option to the default value.

Figure 31: Edit Model Template

Click on Save when done. The model template will be displayed on web UI->PBX->Zero Config-

>Model Templates page.

Click on to delete the model template or click on

“Delete Selected Templates” to delete multiple selected templates at once.

Click on “Toggle Selected Template(s)” to toggle the status between enabled/disabled for the selected model templates.

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DEVICE CONFIGURATION

On web GUI, page PBX->Zero Config->Zero Config, users could create new device, delete existing device(s), make special configuration for a single device, or send NOTIFY to existing device(s).

CREATE NEW DEVICE

Users could also directly create a new device and configure basic settings before the device is discovered by the UCM6100. Once the device is plugged in, it can then be discovered and provisioned by the

UCM6100.

Click on "Create New Device" and the following dialog will show. Follow the steps below to create the configurations for the new device.

1. Firstly select a model for the device to be created and enter its MAC address, IP address and firmware version (optional) in the corresponding field.

2. Basic settings will show a list of settings based on the model selected in step 1. Users could assign extensions to accounts, assign functions to Line Keys and Multiple-Purposed Keys if supported on the selected model.

3. C lick on “Create New Device” to save the configuration for this device.

Firmware Version 1.0.7.10

Figure 32: Create New Device

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MANAGE DEVICES

The device manually created or discovered from Auto Discover will be listed in the web UI->PBX->Zero

Config->Zero Config page. Users can see the devices with their MAC address, IP address, vendor, model and etc.

Figure 33: Manage Devices

Click on to edit the device configuration.

A new dialog will be displayed for the users to configure

“Basic” settings and “Advanced” settings.

“Basic” settings have the same configurations as displayed when manually creating a new device, i.e., account, line key and MPK settings;

“Advanced” settings allow users to configure more details in a five-level structure.

A preview of the

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

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Figure 34: Edit Device

(1) Global Policy

This is the lowest level configuration. The global policy configured in web UI->PBX->Zero

Config->Global Policy will be applied here. Clicking on “Modify Global Policy” to redirect to page

PBX->Zero Config->Global Policy.

(2) Global Templates

Select a global template to be used for the device and click on to add. Multiple global templates can be selected and users can arrange the priority by adjusting orders via and .

All the selected global templates will take effect. If the same option exists on multiple selected global templates, the value in the template with higher priority will override the one in the template with lower priority. Click on to remove the global template from the selected list.

(3) Default Model Template

Default Model Template will be applied to the devices of this model. Default model template can be configured in model template under web UI->PBX->Zero Config->Model Templates page.

Please see default model template option in

[Table 30: Create New Model Template]

.

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(4) Model Templates

Select a model template to be used for the device and click on to add. Multiple global templates can be selected and users can arrange the priority by adjusting orders via and .

All the selected model templates will take effect. If the same option exists on multiple selected model templates, the value in the template with higher priority will override the one in the template with lower priority. Click on to remove the model template from the selected list.

(5) Customize Device Settings

This is the highest level configuration for the device. Click on “Modify Customize Device Settings” and following dialog will show.

Figure 35: Edit Customize Device Settings

Scroll down in the dialog to view and edit the device-specific options. If the users would like to add more options which are not in the predefined list, click on “Add New Field” to add a P value number and the value to the configuration. The following figure shows setting P value “1362” to “en”, which means the display language on the LCD is set to English. For P value information of different models, please refer to configuration template here http://www.grandstream.com/support/tools .

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Figure 36: Add P Value in Customize Device Settings

After the above configurations, save the changes and go back to web UI->PBX->Zero Config->Zero

Config page. Users could then click on to send NOTIFY to the SIP end point device and trigger the provisioning process. The device will start downloading the generated configuration file from the URL contained in the NOTIFY message.

Figure 37: Device List in Zero Config

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

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SAMPLE APPLICATION

Assuming in a small business office where there are 8 GXP2140 phones used by customer support and 1

GXV3275 phone used by customer support supervisor. 3 of the 8 customer support members speak

Spanish and the rest speak English. We could deploy the following configurations to provisioning the office phones for the customer support team.

1. Go to web GUI->PBX->Zero Config->Auto Provision Settings

, select “Enable Zero Config”.

2. Go to web GUI->PBX->Zero Config->Global Policy, configure Date Format, Time Format and

Firmware Source as follows.

Figure 38: Zero Config Sample - Global Policy

3. Go to web GUI->PBX->Zero Config->Model Templates, create a new model template

“English

Support Template” for GXP2140. Add option “Language” and set it to “English”. Then select the option

“Default Model Template” to make it the default model template.

4. Go to web GUI->PBX->Zero Config->Model Templates, create another model template

“Spanish

Support Template” for GXP2140. Add option “Language” and set it to “Español”.

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5. After 9 devices are powered up and connected to the LAN network, use “Auto Discover” function or

“Create New Device” function to add the devices to the device list on web UI->PBX->Zero Config-

>Zero Config.

6. On web GUI->PBX->Zero Config->Zero Config page, users could identify the devices by their MAC addresses or IP addresses displayed on the list. Click on to edit the device settings.

7. For each of the 5 phones used by English speaking customer support, in “Basic” settings select an available extension for account 1 an d click on “Save”. Then click on “Advanced” settings tab to bring up the following dialog. Users will see the English support template is applied since this is the default model template. A preview of the device settings will be listed on the right side.

Figure 39: Zero Config Sample - Device Preview 1

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

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Figure 40: Zero Config Sample - Device Preview 2

Select “Spanish Support Template” in

Model Template

”. The preview of the device settings is displayed on the right side and we can see the language is set to “Español” since Model Template has the higher priority for the option “Language”, which overrides the value configured in default model template.

9. For the GXV3275 used by the customer support supervisor, select an available extension for account

1 on “Basic” settings and click on “Save”. Users can see the preview of the device configuration in

“Advanced” settings. There is no model template configured for GXV3275.

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Figure 41: Zero Config Sample - Device Preview 3

10.

Click on “Apply Changes” to apply saved changes.

11. On the web UI->PBX->Zero Config->Zero Config page, click on to send NOTIFY to trigger the device to download config file from UCM6100.

Now all the 9 phones in the network will be provisioned with an unique extension registered on the

UCM6100. 3 of the phones will be provisioned to display Spanish on LCD and the other 5 will be provisioned to display English on LCD. The GXV3275 used by the supervisor will be provisioned to use the default language on LCD display since it’s not specified in the global policy.

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EXTENSIONS

CREATE NEW USER

CREATE NEW SIP EXTENSION

To manually create new SIP user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on

"Create New User"->"Create New SIP Extension" and a new dialog window will show for users to fill in the extension information. The configuration parameters are as follows.

Table 31: 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.

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CFU Time Condition

Call Forward No Answer

CFN Time Condition

Call Forward Busy

CFB Time Condition

Firmware Version 1.0.7.10

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time condition are

“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is deactivated. The default setting is deactivated.

Select time condition for Call Forward No Answer. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

Configure the Call Forward Busy target number. If not configured, the

Call Forward Busy feature is deactivated. The default setting is deactivated.

Select time condition for Call Forward Busy. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific

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Ring Timeout

Auto Record

Skip Voicemail Password

Verification

Support Hot-Desking Mode

Disable This Extension time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

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.

If selected, this extension will be disabled on the UCM6100.

Note:

The disabled extension still exists on the PBX but can’t be used on the end device.

Select which Music On Hold class to suggest to extension when putting it on hold.

Music On Hold

User Settings

First Name

Last Name

Email Address

Language

Firmware Version 1.0.7.10

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.

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SIP Settings

NAT

Can Reinvite

DTMF Mode

Insecure

Enable Keep-alive

Keep-alive Frequency

Auth ID

TEL URI

Other Settings

Firmware Version 1.0.7.10

To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under web GUI->PBX-

>Internal Options->Language.

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 extension number will be used for authentication.

If the end device/phone has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.

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SRTP

Fax Detection

Skip Trunk Auth

Dial Trunk Password

Strategy

Codec Preference

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.

Note:

If enabled, Fax Pass-through cannot be used.

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

Configure personal password when making outbound calls via trunk.

This option controls how the extension can be used on devices within different types of network.

Allow All

Device in any network can register this extension.

Local Subnet Only

Only the user in specific subnet can register this extension. Up to three subnet addresses can be specified.

A Specific IP Address

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

The default setting is "Allow All".

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

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

Specific Time

Time Condition Click on add Time Condition to configure specific time for this extension.

CREATE NEW IAX EXTENSION

To manually create new IAX user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on

"Create New User"->"Create New IAX Extension" and a new dialog window will show for users to fill in the extension information. The configuration parameters are as follows.

Table 32: IAX Extension Configuration Parameters

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General

Extension

CallerID Number

Permission

SIP/IAX Password

Enable Voicemail

Voicemail Password

Call Forward Unconditional

CFU Time Condition

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.

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time condition are

“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

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Call Forward No Answer

CFN Time Condition

Call Forward Busy

CFB Time Condition

Ring Timeout

Firmware Version 1.0.7.10

Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is deactivated. The default setting is deactivated.

Select time condition for Call Forward No Answer. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

Holiday”, “Out of Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

Configure the Call Forward Busy target number. If not configured, the

Call Forward Busy feature is deactivated. The default setting is deactivated.

Select time condition for Call Forward Busy. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

Configure the 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.

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Auto Record

Skip Voicemail Password

Verification

Disable This Extension

Music On Hold

User Settings

First Name

Last Name

Email Address

Language

IAX Settings

Max Number of Calls

Require Call Token

Other Settings

SRTP

Fax Detection

Firmware Version 1.0.7.10

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 selected, this IAX extension will be disabled on the UCM6100.

Note:

The disabled exte nsion still exists on the PBX but can’t be used on the end device.

Select which Music On Hold class to suggest to extension when putting it on hold.

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.

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.

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

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Skip Trunk Auth

Dial Trunk Password

Strategy

Codec Preference

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.

Note:

If enabled, Fax Pass-through cannot be used.

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

Configure personal password when making outbound calls via trunk.

This option controls how the extension can be used on devices within different types of network.

Allow All

Device in any network can register this extension.

Local Subnet Only

Only the user in specific subnet can register this extension. Up to three subnet addresses can be specified.

A Specific IP Address

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

The default setting is "Allow All".

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

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

Specific Time

Time Condition Click on add Time Condition to configure specific time for this extension.

CREATE NEW FXS EXTENSION

To manually create new FXS user, go to Web GUI->PBX->Basic/Call Routes->Extensions. Click on

"Create New User"->"Create New FXS Extension" and a new dialog window will show for users to fill in the extension information. The configuration parameters are as follows.

Table 33: FXS Extension Configuration Parameters

General

Extension

Analog Station

The extension number associated with the user.

Select the FXS port to be assigned for this extension.

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CallerID Number

Permission

Enable Voicemail

Voicemail Password

Call Forward Unconditional

CFU Time Condition

Call Forward No Answer

CFN Time Condition

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.

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.

Select time condition for Call Forward Unconditional. CFU takes effect only during the selected time condition. The available time condition are

“Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

Configure the Call Forward No Answer target number. If not configured, the Call Forward No Answer feature is deactivated. The default setting is deactivated.

Select time condition for Call Forward No Answer. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of

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Call Forward Busy

CFB Time Condition

Ring Timeout

Auto Record

Skip Voicemail Password

Holiday”, “Out of Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

Configure the Call Forward Busy target number. If not configured, the

Call Forward Busy feature is deactivated. The default setting is deactivated.

Select time condition for Call Forward Busy. The available time condition are “Office Time”, “Out of Office Time”, “Holiday”, “Out of Holiday”, “Out of

Office Time or Holiday” and “Specific”.

Note:

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

Specific time can be configured on the bottom of the extension configuration dialog. Scroll down the add Time Condition for specific time.

Office Time and Holiday could be configured on page Settings-

>Time Settings->Office Time/Holiday page.

Configure the 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

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Verification

Disable This Extension

Music On Hold

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

Firmware Version 1.0.7.10 skipped. If enabled, this would allow one-button voicemail access. By default this option is disabled.

If selected, this FXS extension will be disabled on the UCM6100.

Note:

The disabled extension still exists on the PBX but can’t be used on the end device.

Select which Music On Hold class to suggest to extension when putting it on hold.

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.

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 hookflash 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 hookflash must remain unpressed for the PBX to consider the event as a valid

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Enable Polarity Reversal

Echo Cancellation

3-Way Calling

Send CallerID After

Other Settings

Fax Detection 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 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.

Note:

If enabled, Fax Pass-through cannot be used.

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

Configure personal password when making outbound calls via trunk.

Click on add Time Condition to configure specific time for this extension.

Skip Trunk Auth

Dial Trunk Password

Specific Time

Time Condition

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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 34: Batch Add SIP 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.

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 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.

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

Configure Voicemail password (digits only) for the users.

User Random Password.

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

Use Extension as Password.

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

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the

UCM6100, which can be configured in the global ring timeout setting under web GUI->Internal Options->IVR Prompt: General Preference.

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Auto Record

Skip Voicemail Password

Verification

SIP Settings

NAT

Can Reinvite

DTMF Mode

Insecure

Enable Keep-alive

Keep-alive Frequency

Firmware Version 1.0.7.10

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.

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.

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.

The default setting is 60 seconds.

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TEL URI

If the end device/phone has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.

Other Settings

SRTP

Fax Detection

Strategy

Skip Trunk Auth

Codec Preference

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

Enable to detect Fax signal from the user/trunk during the call and send the received Fax to the Email address configured for this extension. If no

Email address can be found for the user, send the received Fax to the default Email address in Fax setting page under web GUI->PBX-

>Internal Options->Fax/T.38.

Note:

If enabled, Fax Pass-through cannot be used.

This option controls how the extension can be used on devices within different types of network.

Allow All

Device in any network can register this extension.

Local Subnet Only

Only the user in specific subnet can register this extension. Up to 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.722, G.729, G.723, ILBC,

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

BATCH ADD IAX EXTENSIONS

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

IAX Extensions".

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General

Start Extension

Create Number

Permission

Enable Voicemail

SIP/IAX Password

Voicemail Password

Ring Timeout

Auto Record

Firmware Version 1.0.7.10

Table 35: Batch Add IAX Extension Parameters

Configure the starting extension number of the batch of extensions to be added.

Specify the number of extensions to be added. The default setting is 5.

Assign permission level to the user. The available permissions are

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

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.

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

Configure Voicemail password (digits only) for the users.

User Random Password.

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

Use Extension as Password.

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

Configure the number of seconds to ring the user before the call is forwarded to voicemail (voicemail is enabled) or hang up (voicemail is disabled). If not specified, the default ring timeout is 60 seconds on the

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

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Skip Voicemail Password

Verification

IAX Settings

Max Number of Calls

Require Call Token

Other Settings

SRTP

Fax Detection

Strategy

Skip Trunk Auth

Codec Preference

Firmware Version 1.0.7.10 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.

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".

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.

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.722, G.729, G.723, ILBC,

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

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EDIT EXTENSION

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

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

Status

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

Green: Free

Blue: Ringing

Yellow: In Use

Grey: Unavailable (the extension is not registered or disabled on the PBX)

Edit single extension

Click on to start editing the extension parameters.

Reboot the user

Click on to send NOTIFY reboot event to the device which has an UCM6100 extension already registered. To successfully reboot the user, "Zero Config" needs to be enabled on the UCM6100 web

GUI->PBX->Zero Config->Auto Provisioning Settings.

Delete single extension

Click on to delete the extension. Or select the checkbox of the extension and then click on "Delete

Selected Extensions".

Modify selected extensions

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

Delete selected extensions

Select the checkbox for the extension(s). Then click on "Delete Selected Extensions" to delete the extension(s).

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EXPORT EXTENSIONS

The extensions configured on the UCM6100 can be exported to csv format file with selected technology

"SIP", "IAX" or "FXS". Click on "Export Extensions" button and select technology in the prompt below.

Figure 42: Export Extensions

The exported csv file can be serve as a template for users to fill in desired extension information to be imported to the UCM6100.

IMPORT EXTENSIONS

The capability to import extensions to the UCM6100 provides users flexibility to batch add extensions with similar or different configuration quickly into the PBX system.

1. Export extension csv file from the UCM6100 by clicking on "Export Extensions" button.

2. Fill up the extension information you would like in the exported csv template.

3. Click on "Import Extensions" button. The following dialog will be prompted.

Figure 43: Import Extensions

4. Select the option in "On Duplicate Extension" to define how the duplicate extension(s) in the imported

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csv file should be treated by the PBX.

Skip: Duplicate extensions in the csv file will be skipped. The PBX will keep the current extension information as previously configured without change.

Delete and Recreate: The current extension previously configured will be deleted and the duplicate extension in the csv file will be loaded to the PBX.

Update Information: The current extension previously configured in the PBX will be kept.

However, if the duplicate extension in the csv file has different configuration for any options, it will override the configuration for those options in the extension.

5. Click on to select csv file from local directory in the PC.

6. Click on "Save" to import the csv file.

7. Click on "Apply Changes" to apply the imported file on the UCM6100.

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TRUNKS

ANALOG TRUNKS

Go to Web GUI->PBX->Basic/Call Routes->Analog Trunks to add and edit analog trunks.

Click on "Create New Analog Trunk" to add a new analog trunk.

Click on to edit the analog trunk.

Click on to delete the analog trunk.

ANALOG TRUNK CONFIGURATION

The analog trunk options are listed in the table below.

Table 36: Analog Trunk Configuration Parameters

Channels

Trunk Name

SLA Mode

Barge Allowed

Hold Access

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.

Enable this option to satisfy two primary use cases, which include emulating a simple key system and creating shared extensions on a

PBX. Enable SLA Mode will disable polarity reversal.

The barge option specifies whether or not other stations are allowed to join a call in progress on this trunk. If enabled, the other stations can press the line button to join the call. The default setting is Yes.

The h old option specifies hold permissions for this trunk. If set to “Open”, any station can place this trunk on hold and any other station is allowed to retrieve the call. If set to “Private”, only the station that places the call on hold can retrieve the call. The default setting is Yes.

Advanced Options

Enable Polarity Reversal

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

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Polarity on Answer Delay

Current Disconnect Threshold

(ms)

Ring Timeout

RX Gain

TX Gain

Use CallerID

Fax Detection

Caller ID Scheme

FXO Dial Delay(ms)

Auto Record

Disable This Trunk

Maximum Number of Call

Lines

Tone Settings

Busy Detection 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.

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

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

Note:

If enabled, Fax Pass-through cannot be used.

Select the Caller ID scheme for this trunk. The default setting is

“Bellcore/Telcordia”.

Configure the time interval between off-hook and first dialed digit for outbound calls.

Enable automatic recording for the calls using this trunk. The default setting is disabled. The recording files can be accessed under web GUI-

>CDR->Recording Files.

If selected, the trunk will be disabled.

Configure the maximum number of current calls over the trunk at the same time. The default setting is 0, which mean no limit.

Busy Detection is used to detect far end hangup or for detecting busy

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Busy Tone Count

Congestion Detection

Congestion Count

Tone Country

Busy Tone

Congestion Tone

PSTN Detection

Firmware Version 1.0.7.10 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]];

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

Frequencies Range: [0, 4000)

Busy Level Range: (-300, 0)

Cadence Range: [0, 16383].

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

Default value: [email protected],[email protected],c=500/500

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

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

Frequencies Range: [0, 4000)

Busy Level Range: (-300, 0)

Cadence Range: [0, 16383].

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

Default value: [email protected],[email protected],c=250/250

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

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".

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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 44: UCM6100 FXO Tone Settings

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

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Figure 45: 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 46: UCM6100 PSTN Detection: Auto Detect

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

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Figure 47: UCM6100 PSTN Detection: Semi-Auto Detect

Detect Model: Semi-auto Detect.

Source Channel: The source channel to be detected.

Destination Number: The number to be dialed for detecting. This number could be a cell phone number or other PSTN number that can be reached from the source channel PSTN number.

5. Click "Detect" to start detecting. The source channel will initiate a call to the destination number. For

"Auto Detect", the call will be automatically answered. For "Semi-auto Detect", the UCM6100 web

GUI will display prompt to notify the user to answer or hang up the call to finish the detecting process.

6. Once done, the detected result will show. Users could save the detecting result as the current

UCM6100 settings.

Table 37: PSTN Detection for Analog Trunk

Detect Model

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

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Source Channel

Destination Channel

Destination Number 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.

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

Click on to start LDAP Sync.

Click on to delete the VoIP trunk.

The VoIP trunk options are listed in the table below.

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Table 38: SIP Trunk Configuration Parameters

Create New SIP Trunk

Type

Provider Name

Host Name

Keep Trunk CID

Disable This Trunk

TEL URI

Need Registration

Username

Password

Auth ID

Outbound Proxy

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 provider’s server of the trunk.

If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No".

If selected, the trunk will be disabled.

If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.

Select whether the trunk needs to register on the external server or not when "Register SIP Trunk" type is selected. The default setting is No.

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

"Register SIP Trunk" type is selected.

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

"Register SIP Trunk" is selected.

Enter the Authentication ID for "Register SIP Trunk" type.

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

Trunk" type.

Auto Record

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

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.

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

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Keep Trunk CID

Disable This Trunk

TEL URI

Caller ID

CallerID Name

Codec Preference

Auto Record

DID Mode

DTMF Mode

Firmware Version 1.0.7.10

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.

If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No".

If selected, the trunk will be disabled.

If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.

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

When making outgoing calls, the following rules are used to determine 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.

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.

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

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

Configure the default DTMF mode when sending DTMF on this trunk.

Default: The global setting of DTMF mode will be used. The global

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Enable Qualify

Qualify Timeout

Qualify Frequency

Maximum Number of Call

Lines

Fax Detection

SRTP

Sync LDAP Enable

Sync LDAP Password

Sync LDAP Port

LDAP Outbound Rule

LDAP Dialed Prefix

Firmware Version 1.0.7.10 setting for DTMF Mode setting is under web UI->PBX->SIP

Settings->ToS.

RFC2833: Send DTMF using RFC2833.

Info: Send DTMF using SIP INFO message.

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

Auto: Send DTMF using RFC2833 if offered. Otherwise, inband will be used.

If enabled, the UCM6100 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is

“No”.

When

“Enable Qualify” option is set to “Yes”, configure the timeout (in ms) 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.

Configure the maximum number of current calls over this trunk at the same time. The default setting is 0, which means no limit.

Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web

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

Note:

If enabled, Fax Pass-through cannot be used.

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.

Specify an outbound rule for LDAP sync feature. The UCM6100 will automatically modify the remote contacts by adding prefix parsed from this rule.

Specify the prefix for LDAP sync feature. The UCM6100 will

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automatically modify the remote contacts by adding this prefix.

Register SIP Trunk Configuration Parameters

Provider 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.

Host Name

Configure the IP address or URL for the VoIP provider server of the trunk.

Transport

Keep Trunk CID

Disable This Trunk

TEL URI

Need Registration

Username

Password

Auth ID

Codec Preference

From Domain

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.

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

If selected, the trunk will be disabled.

If the trunk has an assigned PSTN telephone number, this field should be set to "User=Phone". Then a "User=Phone" parameter will be attached to the Request-Line and TO header in the SIP request to indicate the E.164 number. If set to "Enable", "Tel:" will be used instead of "SIP:" in the SIP request. The default setting is disabled.

Configure whether to register the trunk to the external server or not. The default setting is Yes.

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.

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 the actual domain name where the extension comes from.

This can be used to override the From Header.

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From User

Outbound Proxy Support

Outbound Proxy

Auto Record

DID Mode

DTMF Mode

Enable Qualify

Qualify Timeout

Qualify Frequency

Maximum Number of Call

Lines

Fax Detection

Firmware Version 1.0.7.10

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.

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.

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

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

Configure the default DTMF mode when sending DTMF on this trunk.

Default: The global setting of DTMF mode will be used. The global setting for DTMF Mode setting is under web UI->PBX->SIP

Settings->ToS.

RFC2833: Send DTMF using RFC2833.

Info: Send DTMF using SIP INFO message.

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

Auto: Send DTMF using RFC2833 if offered. Otherwise, inband will be used.

If enabled, the UCM6100 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No".

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

The maximum number of concurrent calls using the trunk. The default settings 0, which means no limite.

Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web

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

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Note:

If enabled, Fax Pass-through cannot be used.

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

Table 39: IAX Trunk Configuration Parameters

Create New IAX Trunk

Type

Provider Name

Host Name

Keep Trunk CID

Username

Password

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 provider’s server of the trunk.

If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No".

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

"Register IAX Trunk" type is selected.

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

"Register IAX Trunk" type is selected.

If selected, the trunk will be disabled. Disable This Trunk

Peer IAX Trunk Configuration Parameters

Provider 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.

Host Name

Keep Trunk CID

Disable This Trunk

Caller ID

Configure the IP address or URL for the VoIP provider server of the trunk.

If enabled, the trunk CID will not be overridden by extension's CID when the extension has CID configured. The default setting is "No".

If selected, the trunk will be disabled.

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

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

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CallerID Name

Codec Preference

Enable Qualify

Qualify Timeout

Qualify Frequency

Maximum Number of Call

Lines

Fax Detection

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.

If enabled, the UCM6100 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No".

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

The maximum number of concurrent calls using the trunk. The default settings 0, which means no limit.

Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web

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

Note:

If enabled, Fax Pass-through cannot be used.

Register IAX Trunk Configuration Parameters

Provider 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.

Host Name

Configure the IP address or URL for the VoIP provider server of the trunk.

Keep Trunk CID

Disable This Trunk

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

If selected, the trunk will be disabled.

Caller ID

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.

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CallerID Name

Username

Password

Codec Preference

Enable Qualify

Qualify Timeout

Qualify Frequency

Maximum Number of Call

Lines

Fax Detection

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.

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

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

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.

If enabled, the UCM6100 will regularly send SIP OPTIONS to the device to check if the device is still online. The default setting is "No".

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

The maximum number of concurrent calls using the trunk. The default settings 0, which means no limited.

Enable to detect Fax signal from the trunk during the call and send the received Fax to the default Email address in Fax setting page under web

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

Note:

If enabled, Fax Pass-through cannot be used.

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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 UCM6100:

1. To setup DOD go to UCM6100 web GUI->PBX->Basic/Call Routes->VoIP Trunks page.

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

3. Click "Create a new DOD" to begin your DOD setup

4. For "DOD Number" enter one of the numbers (DIDs) from your SIP trunk provider. In the example above Company ABC received 4 DIDs from their provider. ABC will enter in the number for the CEO's direct line.

5. Select an extension from the "Available Extensions" list. Users have the option of selecting more than one extension. In this case, Company ABC would select the CEO's extension. After making the selection, click on the button to move the extension(s) to the "Selected Extensions" list.

Firmware Version 1.0.7.10

Figure 48: DOD extension selection

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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 49: Edit DOD

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SLA STATION

Starting from 1.0.7.10, the UCM6100 supports SLA that allows mapping the key with LED on a multi-line phone to different external lines. When there is an incoming call and the phone starts to ring, the LED on the key will flash in red and the call can be picked up by pressing this key. This allows users to know if the line is occupied or not. The SLA function on the UCM6100 is similar to BLF but SLA is used to monitor external line i.e., analog trunk on the UCM6100. Users could configure the phone with BLF mode on the

MPK to monitor the analog trunk status or press the line key pick up call from the analog trunk on the

UCM6100.

CREATE/EDIT SLA STATION

SLA Station can be configured on web GUI->PBX->Basic/Call Routes->SLA Station.

Figure 50: SLA Station

Click on

“Create New SLA Station” to add a SLA Station.

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

Click on to delete the SLA Station.

Station Name

Station

Available SLA Trunks

Selected SLA Trunks

Table 40: SLA Station Configuration Parameters

Configure a name to identify the SLA Station.

Specify a SIP extension as a station that will be using SLA.

Existing Analog Trunks with SLA Mode enabled will be listed here.

Select a trunk for this SLA from the Available SLA Trunks list. Click on

to arrange the order. If there are multiple trunks selected, when there are calls on those trunks at the same time, pressing the LINE key on the phone will pick up the call on the first trunk here.

SLA Station Options

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Ring Timeout

Ring Delay

Hold Access

Configure the time (in seconds) to ring the station before the call is considered unanswered. No timeout is set by default. If set to 0, there will be no timeout.

Configure the time (in seconds) for delay before ringing the station when a call first coming in on the shared line. No delay is set by default. If set to 0, there will be no delay.

This option defines the competence of the hold action for one particular trunk. If set to “open”, any station could hold a call on that trunk or resume one held session; if set to “private”, only the station that places the trunk call on hold could resume the session. The default setting is

“open”.

SAMPLE CONFIGURATION

1. On the UCM6100, go to web UI->Basic/Call Routes->Analog Trunks page. Create analog trunk or edit the existing analog trunk. Make sure “SLA Mode” is enabled for the analog trunk. Once enabled, this analog trunk will be only available for the SLA stations created under web UI->Basic/Call

Routes->SLA Station page.

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Figure 51: Enable SLA Mode for Analog Trunk

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

Figure 52: Analog Trunk with SLA Mode Enabled

2. On the UCM6100, go to web UI->Basic/Call Routes->SLA Station page, click on “Create New SLA

Station”. Please refer to section

[CREATE/EDIT SLA STATION]

for the configuration parameters. Users can create one or more SLA stations to monitor the analog trunk. The following figure shows two stations, 1002 and 1005, are configured to be associated with SLA trunk

“fxo1”.

Firmware Version 1.0.7.10

Figure 53: SLA Example - SLA Station

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3. On the SIP phone 1, configure to register UCM6100 extension 1002. Configure the MPK as BLF mode and the value must be set to “extension_trunkname”, which is 1002_fxo1 in this case.

4. On the SIP phone 2, configure to register UCM6100 extension 1005. Configure the MPK as BLF mode and value must be set to “extension_trunkname”, which is 1005_fxo1 in this case.

Figure 54: SLA Example - MPK Configuration

Now the SLA station is ready to use. The following functions can be achieved by this configuration.

Making an outbound call from the station/extension, using LINE key

When the extension is in idle state, pressing the line key for this extension on the phone to off hook.

Then dial the station’s extension number, for example, dial 1002 on phone 1 (or dial 1005 on phone

2), to hear the dial tone. Then the users could dial external number for the outbound call.

Making an outbound call from the station/extension, using BLF key

When the extension is in idle state, pressing the MPK and users could dial external numbers directly.

Answering call using LINE key

When the station is ringing, pressing the LINE key to answer the incoming call.

Barging-in active call using BLF key

When there is an active call between an SLA station and an external number using the SLA trunk, other SLA stations monitoring the same trunk could join the call by pressing the BLF key if “Barge

Allowed” is enabled for the analog trunk.

Hold/Unhold using BLF key

If the external line is previously put on hold by an SLA station, another station that monitors the same

SLA trunk could unhold the call by pressing the BLF key i f “Hold Access” is set to “open” on the analog trunk and the SLA station.

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CALL ROUTES

OUTBOUND ROUTES

In the UCM6100, an outgoing calling rule pairs an extension pattern with a trunk used to dial the pattern.

This allows different patterns to be dialed through different trunks (e.g., "Local" 7-digit dials through a FXO while "Long distance" 10-digit dials through a low-cost SIP trunk). Users can also set up a failover trunk to be used when the primary trunk fails.

Go to Web GUI->PBX->Basic/Call Routes->Outbound Routes to add and edit outbound rules.

Click on "Create New Outbound Rule" to add a new outbound route.

Click on to edit the outbound route.

Click on to delete the outbound route.

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 41: Outbound Route Configuration Parameters

Calling Rule Name

Pattern

Password

Call Duration Limit

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.

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

Enable to configure the maximum duration for the call using this outbound route.

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Maximum Call Duration

Warning Time

Warning Repeat Interval

Privilege Level

Enable Filter on Source Caller

ID

Configure the maximum duration of the call (in seconds). The default setting is 0, which means no limite.

Configure the warning time for the call using this outbound route. If set to x seconds, the warning tone will be played to the caller when x seconds are left to end the call.

Configure the warning repeat interval for the call using this outbound route. If set to x seconds, the warning tone will be played every x seconds after the first warning.

Select privilege level for the outbound rule.

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

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

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

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

The default setting is "Disable". Please be aware of the potential security risks when using "Internal" level, which means all users can use this outbound rule to dial out from the trunk.

When enabled, users could specify extensions allowed to use this outbound route. "Privilege Level" is automatically disabled if using

"Enable Filter on Source Caller ID".

The following two methods can be used at the same time to define the extensions as the source caller ID.



Select available extensions/extension groups from the left to the right. This allows users to specify arbitrary single extensions available in the PBX.



Custom Dynamic Route: define the pattern for the source caller ID.

This allows users to define extension range instead of selecting them one by one.

All patterns are prefixed with the "_".

Special characters:

X: Any Digit from 0-9.

Z: Any Digit from 1-9.

N: Any Digit from 2-9.

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

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

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

Send This Call Through Trunk

Use Trunk Select the trunk for this outbound rule.

Strip

Prepend

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.

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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 42: Inbound Rule Configuration Parameters

Trunks

DID Pattern

Privilege Level

Default Destination

Select the trunk to configure the inbound rule.

All patterns are prefixed with the "_".

Special characters:

X: Any Digit from 0-9.

Z: Any Digit from 1-9.

N: Any Digit from 2-9.

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

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

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

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.

Configure the privilege level for this inbound route.

Select the default destination for the inbound call.

Extension

Voicemail

Conference Room

Queue

Ring Group

Paging/Intercom

Voicemail Group

Fax

DISA

IVR

Dial By Name

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Strip

Prepend Trunk Name

Dial Trunk

DID Destination

Alert-Info

Time Condition

Time Conditions

Destination

Firmware Version 1.0.7.10

External Number

By DID

When "By DID" is used, the UCM6100 will look for the destination based on the number dialed, which could be local extensions, conference, call queue, ring group, paging/intercom group, IVR, voicemail groups and Fax extension as configured in "DID destination". If the dialed number matches the DID pattern, the call will be allowed to go through.

Configure the number of digits to be stripped from the beginning of the

DID. This option shows up only when "By DID" is selected.

This option shows up only when

“By DID” is selected. If enabled, the trunk name will be prepended to the display name.

This option shows up only when "By DID" is selected. If enabled, the external users dialing in to the trunk via this inbound route can dial outbound call using the UCM6100’s trunk.

This option shows up only when "By DID" is selected. This controls the destination that can be reached by the external caller via the inbound route. The DID destination are:

Extension

Conference

Call Queue

Ring Group

Paging/Intercom Groups

IVR

Voicemail Groups

Fax Extension

Dial By Name

When present in an INVITE request, the Alert-Info header field specifies an alternative ring tone to the UAS.

Select the time condition for the inbound rule.

Select the destination for the inbound call during the specified time condition.

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BLACKLIST CONFIGURATIONS

In the UCM6100, Blacklist is supported for all inbound routes. Users could enable the Blacklist feature and manage the Blacklist by clicking on "Blacklist".

Figure 55: 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 .

To add blacklist number in batch, click on to upload blacklist file in csv format. The supported csv format is as below.

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Figure 56: Blacklist csv File

Note:

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

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CONFERENCE BRIDGE

The UCM6100 supports conference bridge allowing multiple bridges used at the same time:

UCM6102/6104 supports up to 3 conference bridges allowing up to 25 simultaneous PSTN or IP participants.

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 43: 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 conference

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Enable Caller Menu

Record Conference

Quiet Mode

Wait For Admin

Enable User Invite

Announce Callers

Public Mode

Firmware Version 1.0.7.10 call.

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.

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".

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Play Hold Music

Music On Hold

If enabled, the UCM6100 will play Hold music when there is only one user in the conference. The default setting is "No".

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

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

>Music On Hold.

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".

Skip Authentication When

Inviting User via Trunk from

Web GUI

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 57: Conference Invitation From Web GUI

Invite by dialing 0 or 1 during conference call.

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

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

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

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

Note:

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

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

DURING THE CONFERENCE

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

Manage the conference call from Web GUI.

Log in UCM6100 web GUI during the conference call, the participants in each conference bridge will be listed.

1. Click on to kick a participant from the conference.

2. Click on to mute the participant.

3. Click on to lock this conference bridge so that other users cannot join it anymore.

4. Click on to invite other users into the conference bridge.

Manage the conference call from IVR.

If "Enable Caller Menu" is enabled, conference participant can input * to enter the IVR menu for the conference. Please see options listed in the table below.

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Table 44: Conference Caller IVR Menu

6

7

8

1

4

5

Conference Administrator IVR Menu

1 Mute/unmute yourself.

2

3

Lock/unlock the conference bridge.

Kick the last joined user from the conference.

4

5

6

Decrease the volume of the conference call.

Decrease your volume.

Increase the volume of the conference call.

7

8

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.

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To record the conference call, when the conference bridge is in idle, enable "Record Conference" from the conference bridge configuration dialog. Save the setting and apply the change. When the conference call starts, the call will be automatically recorded in .wav format.

The recording files will be listed as below once available. Users could click on to download the recording or click on to delete the recording. Users could also delete all recording files by clicking on

“Delate All Recording Files”, or delete multiple recording files at once by clicking on “Delete Selected

Recording Files

” after selecting the recording files.

Figure 58: Conference Recording

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IVR

CONFIGURE IVR

IVR configurations can be accessed under the UCM6100 Web GUI->PBX->Call Features->IVR. Users could create, edit, view and delete an IVR.

Click on "Create New IVR" to add a new IVR.

Click on to edit the IVR configuration.

Click on to delete the IVR.

Name

Extension

Dial Other Extensions

Dial Trunk

Permission

Welcome Prompt

Digit Timeout

Response Timeout

Table 45: 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 detected within

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Response Timeout Prompt

Invalid Prompt

Response Timeout Repeat

Loops

Invalid Repeat Loops

Language

Key Press Event 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 UCM6100.

To add more languages in the list, please download voice prompt package by selecting "Check Prompt List" under web GUI->PBX-

>Internal Options->Language.

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

The event options are:

Extension

Voicemail

Conference Rooms

Voicemail Group

IVR

Ring Group

Queues

Page Group

Fax

Custom Prompt

Hangup

DISA

Dial By Name

External Number

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CREATE CUSTOM PROMPT

To record new IVR prompt or upload IVR prompt to be used in IVR, click on

“Prompt” next to the

“Welcome Prompt” option and the users will be redirected to Custom Prompt page. Or users could go to

Web GUI->PBX->Internal Options->Custom Prompt page directly.

Figure 59: Click on Prompt to Create IVR Prompt

Once the IVR prompt file is successfully added to the UCM6100, it will be added into the prompt list options for users to select in different IVR scenarios.

RECORD NEW CUSTOM PROMPT

In the UCM6100 Web GUI->PBX->Internal Options->Custom Prompt page, click on

“Record New

Custom Prompt

” and follow the steps below to record new IVR prompt.

Figure 60: Record New Custom Prompt

Specify the IVR file name.

Select the format (GSM or WAV) for the IVR prompt file to be recorded.

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Select the extension to receive the call from the UCM6100 to record the IVR prompt.

Click the “Record” button. A request will be sent to the UCM6100. The UCM6100 will then call the extension for recording the IVR prompt from the phone.

Pick up the call from the extension and start the recording following the voice prompt.

The recorded file will be listed in the IVR Prompt web page. Users could select to re-record, play or delete the recording.

UPLOAD CUSTOM PROMPT

If the user has a prerecorded IVR prompt file, click on “Upload Custom Prompt” in Web GUI->PBX-

>Internal Options->Custom Prompt page to upload the file to the UCM6100. The following are required for the IVR prompt file to be successfully uploaded and used by the UCM6100:

PCM encoded.

16 bits.

8000Hz mono.

In .mp3 or .wav format; or raw/ulaw/alaw/gsm file with .ulaw or .alaw suffix.

File size under 5M.

Figure 61: Upload Custom Prompt

Click on to select audio file from local PC and click on to start uploading. Once uploaded, the file will appear in the Custom Prompt web page.

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LANGUAGE SETTINGS FOR VOICE PROMPT

The UCM6100 supports multiple languages in web GUI as well as system voice prompt. The following languages are currently supported in system voice prompt:

English (United States)

Arabic

Chinese

Dutch

English (United Kingdom)

French

German

Greek

Hebrew

Italian

Polish

Portuguese

Russian

Spanish

Swedish

Turkish

English (United States) and Chinese voice prompts are built in with the UCM6100 already. The other languages provided by Grandstream can be downloaded and installed from the UCM6100 web GUI directly. Additionally, users could customize their own voice prompts, package them and upload to the

UCM6100.

Language settings for voice prompt can be accessed under Web GUI->PBX->Internal Options-

>Language.

DOWNLOAD AND INSTALL VOICE PROMPT PACKAGE

To download and install voice prompt package in different languages from UCM6100 web GUI, click on

"Check Prompt List" button.

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Figure 62: Language Settings For Voice Prompt

A new dialog window of voice prompt package list will be displayed. Users can see the version number

(latest version available V.S. current installed version), package size and options to upgrade or download the language.

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Figure 63: Voice Prompt Package List

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

Figure 64: New Voice Prompt Language Added

A new language option will be displayed after successfully installed. Users then could select it to apply in the UCM6100 system voice prompt or delete it from the UCM6100.

CUSTOMIZE AND UPLOAD VOICE PROMPT PACKAGE

The UCM6100 provides interface from web GUI for users to customize their own voice prompts. Users could directly upload the package from web GUI. For detailed instructions on voice prompt customizing and uploading, please refer to the link below: http://www.grandstream.com/products/ucm_series/UCM6100/documents/UCM6100_voiceprompt_custom ization.zip

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VOICEMAIL

CONFIGURE VOICEMAIL

If the voicemail is enabled for UCM6100 extensions, the configurations of the voicemail can be globally set up and managed under Web GUI->PBX->Call Features->Voicemail.

Table 46: Voicemail Settings

Max Greeting

Dial

‘0’ For Operator

Max Messages Per Folder

Configure the maximum number of seconds for the voicemail greeting.

The default setting is 60 seconds.

If enabled, the caller can press 0 to exit the voicemail application and 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.

Max Message Time

Min Effective Message Time

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

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

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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 65: Voicemail Email Settings

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Table 47: Voicemail Email Settings

Attach Recordings to E-Mail

Template For Voicemail Emails

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

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

The template variables are:

\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.

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Figure 66: Voicemail Group

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Extension

Name

Voicemail Password

Email Address

Voicemail Group Mailboxes

Table 48: 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.

Configure the voicemail password for the users to check voicemail messages.

Configure the Email address for the voicemail group extension.

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

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

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RING GROUP

The UCM6100 supports ring group feature with different ring strategies applied to the ring group members. This section describes the ring group configuration on the UCM6100.

CONFIGURE RING GROUP

Ring group settings can be accessed via Web GUI->PBX->Call Features->Ring Group.

Figure 67: 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.

Ring Group Name

Extension

Ring Group Members

Ring Strategy

Permission

Table 49: Ring Group Parameters

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

– are allowed.

Configure the ring group extension.

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

Select the ring strategy. The default setting is

“Ring in order”.

Ring simultaneously.

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

Ring in order.

Ring the members with the order configured in ring group list. If the first member doesn

’t answer the call, it will stop ringing the first member and start ringing the second member.

Configure the permission level for Ring Group. This permission will take effect when an incoming call is going to be directed to an external number configured as “Default Destination”. The available permissions are "Internal", "Local", "National" and "International" from the lowest level

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Custom Prompt to the highest level.

This option is to set a custom prompt for a ring group to announce to caller. Click on ‘Prompt’, it will direct to the page PBX->Internal Options-

>Custom Prompt, where users could record new prompt or upload prompt files.

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.

Note:

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

Auto Record

Enable Destination

Secret

Email Address

If enabled, calls on this ring group will be automatically recorded. The default setting is No. The recording files can be accessed from web GUI-

>CDR->Recording Files.

If enabled, users could select extension, voicemail, ring group, IVR, call 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:

The password has to be at least 4 characters.

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.

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Figure 68: Ring Group Configuration

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PAGING AND INTERCOM GROUP

The UCM6100 paging and intercom can be used via feature code to a single extension or a paging/intercom group. This sections describes the configuration of paging/intercom group under Web

GUI->PBX->Call Features->Paging/Intercom.

CONFIGURE PAGING/INTERCOM GROUP

Click on "Create New Paging/Intercom Group" to add paging/intercom group.

Name

Extension

Type

Custom Prompt

Page/Intercom Group

Members

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Figure 69: Paging/Intercom Group

Table 50: Paging/Intercom Group Configuration Parameters

Configure paging/intercom group name.

Configure the paging/intercom group extension.

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

This option is to set a custom prompt for a paging/intercom group to announce to caller. Click on ‘Prompt’, it will direct to the page PBX-

>Internal Options->Custom Prompt, where users could record new prompt or upload prompt files.

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

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Click on to edit the paging/intercom group.

Click on to delete the paging/intercom group.

Click on "Paging/Intercom Group Settings" to edit Alert-Info Header. This header will be included in the SIP INVITE message sent to the callee in paging/intercom call.

Figure 70: Page/Intercom Group Settings

The UCM6100 has pre-configured paging/intercom feature code. By default, the Paging Prefix is *81 and the Intercom Prefix is *80. To edit page/intercom feature code, click on "Feature Codes" in the

"Paging/Intercom Group Settings" dialog. Or users could go to Web GUI->PBX->Internal Options-

>Feature Codes directly.

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CALL QUEUE

The UCM6100 supports call queue by using static agents or dynamic agents. This sections describes the configuration of call queue under Web GUI->PBX->Call Features->Call Queue.

CONFIGURE CALL QUEUE

Call queue settings can be accessed via Web GUI->PBX->Call Features->Call Queue.

Figure 71: Call Queue

Click on "Create New Queue" to add call queue.

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

Extension

Name

Strategy

Table 51: 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.

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Music On Hold

Leave When Empty

Dial in Empty Queue

Permission

Dynamic Login Password

Fewest Calls

Ring the agent with the fewest completed calls.

Random

Ring a random agent.

Round Robin

Ring the agents in Round Robin scheduling with memory.

The default setting is "Ring All".

Select the Music On Hold class for the call queue.

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.

Configure the permission level for Call Queue. This permission will take effect when an incoming call is going to be directed to an external number configured as “Default Destination”. The available permissions are "Internal", "Local", "National" and "International" from the lowest level to the highest level.

If enabled, the configured PIN number is required for dynamic agent to

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Ring Time Out

Wrapup Time

Max Queue Length

Report Hold Time

Wait Time

Auto Record

Enable Destination

Default Destination

Agents 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.

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".

If enabled, the calls on the call queue will be automatically recorded. The recording files can be accessed in Queue Recordings under web GUI-

>PBX->Call Features->Call Queue.

If enabled, the incoming call for the call queue will be routed to the destination configured in the next field if none of the agents answers the call after ringing for a time of “Ring Timeout”.

Configure the call destination for the call to be routed to if no agent in this call queue answers the call.

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

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.

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Figure 72: Agent Login Settings

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

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

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

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

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

*84 for "Agent Unpause".

Queue recordings are shown on the Call Queue page. Click on to download the recording file in

.wav format; click on to delete the recording file. To delete multiple recording files by one click, select several recording files to be deleted and click on “Delete Selected Recording Files” or click on

“Delete All Recording Files” to delete all recording files.

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EXTENSION GROUPS

The UCM6100 extension group feature allows users to assign and categorize extensions in different groups to better manage the configurations on the UCM6100. For example, when configuring "Enable

Filter on Source Caller ID", users could select a group instead of each person's extension to assign. This feature simplifies the configuration process and helps manage and categorize the extensions for business environment.

CONFIGURE EXTENSION GROUPS

Extension group can be configured via Web GUI->PBX->Call Features->Extension Groups.

Click on "Create New Extension Group" to create a new extension group.

Click on to edit the extension group.

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

Figure 73: Edit Extension Group

Click on to delete the extension group.

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USE EXTENSION GROUPS

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

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

Figure 74: Select Extension Group in Outbound Route

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PICKUP GROUPS

The UCM6100 supports pickup group feature which allows users to pick up incoming calls for other extensions if they are in the same pickup group, by dialing "Pickup Extension" feature code (by default

*8).

CONFIGURE PICKUP GROUPS

Pickup groups can be configured via Web GUI->PBX->Call Features->Pickup Groups.

Click on "Create New Pickup Group" to create a new pickup group.

Click on to edit the pickup group.

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

Figure 75: Edit Pickup Group

Click on to delete the pickup group.

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MUSIC ON HOLD

Music On Hold settings can be accessed via Web GUI->PBX->Internal Options->Music On Hold. In this page, users could configure music on hold class and upload music files. The "default" Music On Hold class already has 5 audio files defined for users to use.

Figure 76: Music On Hold Default Class

Click on "Create New MOH Class" to add a new Music On Hold class.

Click on to configure the MOH class sort method to be "Alpha" or "Random" for the sound files.

Click on next to the selected Music On Hold class to delete this Music On Hold class.

Click on to select music file from local PC and click on to start uploading. The music file uploaded has to be 8 KHz Mono format with size smaller than 5M.

Click on next to the sound file to delete it from the selected Music On Hold Class.

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Note:

Once the MOH file is deleted, there are two ways to recover the music files.

Users could download the MOH file from this link: http://downloads.asterisk.org/pub/telephony/sounds/releases/asterisk-moh-opsound-wav-2.03.tar.gz

After downloading and unzip the pack, users could then upload the music files to UCM.

Factory reset could also recover the MOH file on the UCM.

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FAX/T.38

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

>PBX->Internal Options->FAX/T.38. The list of received Fax files will be displayed in the same web page for users to view, retrieve and delete.

CONFIGURE FAX/T.38

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

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

Table 52: 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

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

Click on to edit the Fax extension.

Click on to delete the Fax extension.

SAMPLE CONFIGURATION TO RECEIVE FAX FROM PSTN LINE

The following instructions describes how to use the UCM6100 to receive Fax from PSTN line on the Fax machine connected to the UCM6100 FXS port.

1. Connect Fax machine to the UCM6100 FXS port.

2. Connect PSTN line to the UCM6100 FXO port.

3. Go to web GUI->PBX->Analog Trunks page.

4. Create or edit the analog trunk for Fax as below.

Fax Detection: Make sure "Fax Detection" option is set to "No".

Figure 77: Configure Analog Trunk without Fax Detection

5. Go to UCM6100 web GUI->PBX->Basic/Call Routes->Extensions page.

6. Create or edit the extension for FXS port.

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Analog Station: Select FXS port to be assigned to the extension. By default, it's set to "None".

Once selected, analog related settings for this extension will show up in "Analog Settings" section.

Figure 78: 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.

Firmware Version 1.0.7.10

Figure 79: Configure Inbound Rule for Fax

UCM6100 Series IP PBX User Manual Page 165 of 239

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 80: Create Fax Extension

3. Go to UCM6100 web GUI->Basic/Call Routes->Analog Trunks page. Create a new analog trunk.

Please make sure "Fax Detection" is set to "No".

4. Go to UCM6100 web GUI->Basic/Call Routes->Inbound Routes page. Create a new inbound route and set the default destination to the Fax extension.

Firmware Version 1.0.7.10

Figure 81: Inbound Route to Fax Extension

UCM6100 Series IP PBX User Manual Page 166 of 239

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

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FOLLOW ME

Follow Me is a feature on the UCM6100 that allows users to direct calls to other phone numbers and have them ring all at once or one after the other.

Calls can be directed to users’ home phone, office phone, mobile and etc. The calls will get to the user no matter where they are.

To configure follow me:

Click on "Create New Follow Me" and then select an extension to be configured with Follow Me.

Figure 82: Create Follow Me

Click on “Next” to continue editing Follow Me configuration.

Figure 83: Edit Follow Me

Click on “Add Follow Me Number” to add local extensions or external numbers to be called after ringing the extension selected in the first step.

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Once created, it will be displayed on the follow me web page list. Click on to edit the Follow Me configuration. Click on to delete the Follow Me.

The following table shows the Follow Me configuration parameters.

Table 53: Follow Me Settings

Enable

Music On Hold Class

Configure to enable or disable Follow Me for this user.

Configure the Music On Hold class that the caller would hear while tracking the user.

Follow Me Numbers

New Follow Me Number

The added numbers are listed here. Click on to arrange the order.

Click on to delete the number. Click on to add new numbers.

Add a new Follow Me number which could be a ‘Local Extension’ or

‘External Number’. The selected dial plan should have permissions to dial the defined external number.

Dialing Order

Select the order in which the Follow Me destinations will be dialed to reach the user: ring all at once or ring one after the other.

Click on

“Follow Me Options” to enable or disable the options listed in the following table.

Table 54: Follow Me Options

Playback Incoming Status

Message

Record the Caller’s Name

Playback Unreachable Status

Message

If enabled, the PBX will playback the incoming status message before starting the Follow Me steps.

If enabled, th e PBX will record the caller’s name from the phone so it can be announced to the callee in each step.

If enabled, the PBX will playback the unreachable status message to the caller if the callee cannot be reached.

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ONE-KEY DIAL

The UCM6100 supports One-Key Dial that allows users to call a certain destination by pressing one digit

0 to 9 on the keypad. To enable One-Key Dial, on UCM6100 web GUI, go to page PBX->Call Features-

>One-Key Dial. User should first decide a digit used for One-

Key Dial and check the option “Enable

Destination” for the digit. Then select a dial destination from “Default Destination”. The supported destinations include extension, voicemail, conference room, voicemail group, IVR, ring group, call queue, page group, fax, DISA, Dial by Name and external number.

Figure 84: Configure One-Key Dial

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DISA

The UCM6100 supports DISA to be used in IVR or inbound route. Before using it, create new DISA under web GUI->Call Features->DISA.

Click on "Create New IVR" to add a new DISA.

Click on to edit the DISA configuration.

Click on to delete the DISA.

Name

Password

Permission

Firmware Version 1.0.7.10

Figure 85: Create New DISA

Table 55: 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

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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.

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BLF AND EVENT LIST

BLF

The UCM6100 supports BLF monitoring for extensions, ring group, call queue, conference room and parking lot. For example, on the user's phone, configure the parking lot number 701 as the BLF monitored number. When there is a parked call on 701, the LED for this BLF key will light up in red, meaning a call is parked against this parking lot. Pressing this BLF key can pick up the call from this parking lot.

Note:

On the Grandstream GXP series phones, the MPK supports "Call Park" mode, which is normally used to park the call by configuring the MPK number as call park feature code (e.g., 700). Users could also use "Call Park" mode to monitor and pick up the call on this parking lot by configuring the MPK number as parking lot number (e.g., 701).

EVENT LIST

Besides BLF, users can also configure the phones to monitor event list. In this way, both local extensions on the same UCM6100 and remote extensions on the VOIP trunk can be monitored. The event list settings is under web GUI->Call Features->Event List.

Click on "Create New Event List" to add a new event list.

Click on to edit the event list configuration.

Click on to delete the event list.

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URI

Local Extensions

Remote Extensions

Special Extensions

Firmware Version 1.0.7.10

Figure 86: Create New Event List

Table 56: Event List Settings

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/Extension Groups listed on the local

UCM6100 to be monitored in the event list.

If LDAP sync is enabled between the UCM6100 and the peer UCM6100, 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

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Remote extension monitoring works on the UCM6100 via event list BLF, among Peer SIP trunks or

Register SIP trunks (register to each other). Therefore, please properly configure SIP trunks on the

UCM6100 first before using remote BLF feature. Please note the SIP end points need support event list

BLF in order to monitor remote extensions.

When an event list is created on the UCM6100 and remote extensions are added to the list, the

UCM6100 will send out SIP SUBSCIRBE to the remote UCM6100 to obtain the remote extension status.

When the SIP end points registers and subscribes to the local UCM6100 event list, it can obtain the remote extension status from this event list.

Once successfully configured, the event list page will show the status of total extension and subscribers for each event list. Users can also select the event URI to check the monitored extension's status and the subscribers' details.

Note:

To configure LDAP sync, please go to UCM6100 web GUI->PBX->Basic/Call Routes->VoIP Trunk.

You will see "Sync LDAP Enable" option. Once enabled, please configure password information for the remote peer UCM6100 to connect to the local UCM6100. Additional information such as port number, LDAP outbound rule, LDAP Dialed Prefix will also be required. Both the local UCM6100 and remote UCM6100 need enable LDAP sync option with the same password for successful connection and synchronization.

Currently LDAP sync feature only works between two UCM6100s.

(Theoretically) Remote BLF monitoring will work when the remote PBX being monitored is non-

UCM6100 PBX. However, it might not work the other way around depending on whether the non-

UCM6100 PBX supports event list BLF or remote monitoring feature.

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DIAL BY NAME

Dial By Name is a feature on the PBX that allows caller to search a person by first or last name via his/her phone's keypad. The administrator can define the Dial By Name directory including the desired extensions in the directory and the searching type by "first name" or "last name". After dialing in, the PBX

IVR/Auto Attendant will guide the caller to spell the digits to find the person in the Dial By Name directory.

This feature allows customers/clients to use the guided automatic system to get in touch with the enterprise employees without having to know the extension number, which brings convenience and improves business image for the enterprise.

DIAL BY NAME CONFIGURATION

The administrators can create the dial by name group under web GUI->PBX->Call Features->Dial By

Name.

Figure 87: Create Dial By Name Group

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1. Group Name

Enter the Group Name. This is to identify the Dial By Name group. The Dial By Name group can be used as the destination for inbound route and key pressing event for IVR. The group name defined here will show up in the destination list when configuring IVR and inbound route.

Figure 88: Dial By Name Group In IVR Key Pressing Events

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Figure 89: Dial By Name Group In Inbound Rule

2. Extension

Configure the direct dial extension for the Dial By Name group.

3. Available Extensions/Selected Extensions

Select available extensions from the left side to the right side as the directory for the Dial By Name group.

Only the selected extensions here can be reached by the Dial By Name IVR when dialing into this group.

The extensions here must have a valid first name and last name configured under web GUI->PBX-

>Basic/Call Routes->Extensions in order to be searchable in Dial By Name directory through IVR. By specifying the extensions here, the administrators can make sure unscreened calls will not reach the company employee if he/she doesn't want to receive them directly.

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Figure 90: Configure Extension First Name and Last Name

4. Query Type

Specify the query type. This defines how the caller will need to enter to search the directory.

By First Name: enter the first 3 digits of the first name to search the directory.

By Last Name: enter the first 3 digits of the last name to search the directory.

By Full Name: enter the first 3 digits of the first name or last name to search the directory.

5. Select Type

Specify the select type on the searching result. The IVR will confirm the name/number for the party the caller would like to reach before dialing out

By Order: After the caller enters the digits, the IVR will announce the first matching party's name and number. The caller can confirm and dial out if it's the destination party, or press * to listen to the next matching result if it's not the desired party to call.

By Menu: After the caller enters the digits, the IVR will announce 8 matching results. The caller can press number 1 to 8 to select and call, or press 9 for results in next page.

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CALL FEATURES

The UCM6100 supports call recording, transfer, call forward, call park and other call features via feature code. This section lists all the feature codes in the UCM6100 and describes how to use the call features.

FEATURE CODES

Table 57: UCM6100 Feature Codes

Feature Maps

Blind Transfer

Attended Transfer

Disconnect

Call Park

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

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

Default code: *0.

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

Options

Disable

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

Default code: #72.

Enter the code during active call to park the call.

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Audio Mix Record

Options

Disable

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

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

Allow Caller: Enable the feature code on caller side only.

Allow Callee: Enable the feature code on callee side only.

Allow Both: Enable the feature code on both caller and callee.

DND/Call Forward

Do Not Disturb (DND) Activate

Default code: *77.

Do Not Disturb (DND) Deactivate

Default code: *78.

Call Forward Busy Activate

Default Code: *90.

Enter the code and follow the voice prompt. Or enter the code

Call Forward Busy Deactivate

Call Forward No Answer Activate followed by the extension to forward the call.

Default Code: *91.

Default Code: *92.

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

Call Forward No Answer

Deactivate

Default Code: *93.

Call Forward Unconditional

Activate

Default Code: *72.

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

Call Forward Unconditional

Deactivate

Feature Misc

Default Code: *73.

Feature Code Digits Timeout

Call Park

Default Setting: 1000.

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.

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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

Pickup Extension

Firmware Version 1.0.7.10

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.

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

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Direct Dial Voicemail Prefix 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 an extension, ring group, call queue or trunk, the call will be automatically recorded when there is established call with it. Otherwise, please follow the instructions below to manually record the call.

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

2. After establishing the call, enter the "Audio Mix Record" feature code (by default it's *3) followed by # or SEND to start recording.

3. To stop the recording, enter the "Audio Mix Record" feature code (by default it's *3) followed by # or

SEND again. Or the recording will be stopped once the call hangs up.

4. The recording file can be retrieved under Web GUI->Status->CDR. Click on to play the recording or click on to download the recording file.

Figure 91: Download Recording File From CDR Page

The above recorded call's recording files are also listed under the UCM6100 web GUI->CDR->Recording

Files.

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CALL PARK

The UCM6100 provides call park and call pickup features via feature code.

PARK A CALL

There are two feature codes that can be used to park the call.

Feature Maps->Call Park (Default code #72)

During an active call, press #72 and the call will be parked. Parking lot number (default range 701 to

720) will be announced after parking the call.

Feature Misc->Call Park (Default code 700)

During an active call, initiate blind transfer (default code #1) and then dial 700 to park the call. Parking lot number (default range 701 to 720) will be announced after parking the call.

RETRIEVE THE PARKED CALL

To retrieve the parked call, simply dial the parking lot number and the call will be established. If a parked call is not retrieved after the timeout, the original extension who parks the call will be called back.

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INTERNAL OPTIONS

This section describes internal options that haven't been mentioned in previous sections yet. The settings in this section can be applied globally to the UCM6100, including general configurations, jitter buffer, RTP settings, ports config and STUN monitor. The options can be accessed via Web GUI->PBX->Internal

Options.

INTERNAL OPTIONS/GENERAL

Table 58: 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:



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

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Enable Random Password

Disable Extension Range

Extension Ranges

Firmware Version 1.0.7.10 digits, with a minimum length of 4 digits. Repetitive digits pattern

(such as 0000, 1111, 1234, 2345, and etc), or common digits pattern

(such as 111222, 321321 and etc) are not allowed to be configured as password.



Password for extension registration, web GUI admin login, LDAP and

LDAP sync requires alphanumeric characters containing at least two categories of the following, with a minimum length of 4 characters.

Numeric digits

Lowercase alphabet characters

Uppercase alphabet characters

Special characters

If enabled, random password will be generated when the extension is 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 preconfigured/configured on the UCM6100. The default setting is "No".

Note:

It is recommended to keep the system assignment to avoid inappropriate usage and unnecessary issues.

The default extension range assignment is:

User Extensions: 1000-6299

User Extensions is referring to the extensions created under web UI-

>PBX->Basic/Call Routes->Extensions page.

Pick Extensions: 4000-4999

This refers to the extensions that can be manually picked from end device when being provisioned by the UCM6100. There are two related options in zero config page->Auto Provision Settings, "Pick

Extension Segment" and "Enable Pick Extension". If "Enable Pick

Extension" under zero config settings is selected, the extension list defined in "Pick Extension Segment" will be sent out to the device after receiving the device's request. This "Pick Extension Segment" should be a subset of the "Pick Extensions" range here. This feature is for the GXP series phones that support selecting extension to be provisioned via phone's LCD.

Auto Provision Extensions: 5000-6299

This sets the range for "Zero Config Extension Segment" which is the

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extensions can be assigned on the UCM6100 to provision the end device.

Conference Extensions: 6300-6399

Ring Group Extensions: 6400-6499

Queue Extensions: 6500-6599

Voicemail Group Extensions: 6600-6699

IVR Extensions: 7000-7100

Fax Extensions: 7200-8200

INTERNAL OPTIONS/JITTER BUFFER

Table 59: 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 60: Internal Options/RTP Settings

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

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RTP End

Strict RTP

RTP Checksums

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/PORTS CONFIG

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

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

Figure 92: 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

Firmware Version 1.0.7.10

Figure 93: FXO Ports ACIM Settings

Table 61: Internal Options/Ports 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)".

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FXO Opermode

FXS Opermode

FXS TISS Override

PCMA Override

Boost Ringer

Fast Ringer

Low Power

Ring Detect

FXS MWI Mode

Firmware Version 1.0.7.10

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 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 States of America (USA)".

Configure to enable or disable override Two-Wire Impedance Synthesis

(TISS). The default setting is No.

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

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

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STUN Server

STUN Refresh

INTERNAL OPTIONS/STUN MONITOR

Table 62: Internal Options/STUN Monitor

Configures the IP address or URL of the STUN server to query. If not specified, STUN is disabled. The default setting is stun.ipvideotalk.com.

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.

INTERNAL OPTIONS/PAYLOAD

The UCM6100 payload type for audio codecs and video codes can be configured here.

Table 63: Internal Options/Payload

AAL2-G.726

DTMF

G.721 Compatible

G.726

ILBC

H.264

H.263P

Configure payload type for ADPCM (G.726, 32kbps, AAL2 codeword packing). The default setting is 112.

Configured payload type for DTMF. The default setting is 101.

Configure to enable/disable G.721 compatible. The default setting is Yes.

Configure the payload type for G.726 if "G.721 Compatible" is disabled.

The default setting is 111.

Configure the payload type for ILBC. The default setting is 97.

Configure the payload type for H.264. The default setting is 99.

Configure the payload type for H.263+. The default setting is 100 103.

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IAX SETTINGS

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

IAX SETTINGS/GENERAL

Bind Port

Bind Address

IAX1 Compatibility

No Checksums

Delay Reject

ADSI

Music On Hold Interpret

Music On Hold Suggest

Table 64: 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 65: IAX Settings/Registration

IAX Registration Options

Min Reg Expire

Max Reg Expire

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

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

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IAX Thread Count

IAX Max Thread Count

Auto Kill

Authentication Debugging

Codec Priority

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

Configure the maximum number of IAX threads allowed. The default setting is 100.

If set to "yes", the connection will be terminated if ACK for the NEW 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 66: 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.

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Max Unvalidated Call Numbers Configure the maximum number of unvalidated calls for all IP addresses.

Call Number Limits Configure to limit the number of calls for a give IP address of IP range.

IP or IP Range

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.

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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 67: 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.

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From Domain

Auto Domain

Allow External Domains

Configure the domain in the "From:" header of the SIP message. It may be required by some providers for authentication.

If enabled, the UCM6100 will add local host name and local IP to domain list. The default setting is "No".

If enabled, requests for external domains that are not served by the

UCM6100 will be allowed. The default setting is "Yes".

SIP SETTINGS/MISC

Table 68: 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 69: 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.

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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 70: 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.

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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

Firmware Version 1.0.7.10

Table 71: 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:

192.168.0.0/16

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SIP SETTINGS/TOS

ToS For SIP

ToS For RTP Audio

ToS For RTP Video

Default Incoming/Outgoing

Registration Time

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

Table 72: 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.

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".

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Generate In-Band Ringing

Server User Agent

Send Compact SIP Headers

Add "user=phone" to URI

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.

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".

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STATUS AND REPORTING

PBX STATUS

The UCM6100 monitors the status for Trunks, Extensions, Queues, Conference Rooms, Interfaces and

Parking lot. It presents administrators the real time status in different sections under web GUI->Status-

>PBX Status.

Figure 94: Status->PBX Status

TRUNKS

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

Status

Firmware Version 1.0.7.10

Figure 95: Trunk Status

Table 73: Trunk Status

Display trunk status.

Analog trunk status:

Available

Busy

Unavailable

Unknown Error

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SIP Peer trunk status:

Unreachable: The hostname cannot be reached.

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

Reachable: The hostname can be reached.

SIP Register trunk status:

Registered

Unrecognized Trunk

Display trunk name Trunks

Type

Username

Display trunk Type:

Analog

SIP

IAX

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.

Firmware Version 1.0.7.10

Figure 96: Extension Status

UCM6100 Series IP PBX User Manual Page 201 of 239

Table 74: Extension Status

Status

Extension

Name/Label

Message

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.

Type

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.

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Figure 97: 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 75: 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.

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Figure 98: 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.

Firmware Version 1.0.7.10

Figure 99: UCM6116 Interfaces Status

Table 76: Interface Status Indicators

USB connected.

USB disconnected.

SD Card connected.

SD Card disconnected.

LAN/WAN connected.

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LAN/WAN not configured.

LAN/WAN disconnected.

FXS/FXO connected.

FXS/FXO waiting.

FXS/FXO busy.

FXS/FXO not configured.

FXS/FXO disconnected.

Other operations are also available in interface status section:

Click on "Interfaces Status", the web page will redirect to ports configuration page which can also be accessed via web GUI->PBX->Internal Options->Ports Config.

Click on 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.

Caller ID

Channel

Extension

Timeout

Firmware Version 1.0.7.10

Figure 100: Parking Lot Status

Table 77: Parking Lot Status

Display the caller ID who parks the call.

Display channel for the call park.

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.

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Other operations are also available in parking lot status section:

Click on "Parking Lot", the web page will redirect to feature codes page which can also be accessed via web GUI->PBX->Internal Options->Feature Codes.

Click on to refresh the parking lot status.

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

Click on [ - ] to hide the parking details.

ACTIVITY CALLS

The UCM6100 can monitor the status of active calls in real time. The active calls status can be viewed under web GUI->Status->Active Calls.

The following figure shows extension 1002 is calling 1001. 1001 is ringing.

Figure 101: Status->PBX Status->Activity Calls: Calling

The following figure shows the call between 1001 and 1002 is established.

Firmware Version 1.0.7.10

Figure 102: Status->PBX Status->Activity Calls

UCM6100 Series IP PBX User Manual Page 206 of 239

Click on to refresh the active call status.

Click on to hang up all calls.

Click on to hang up single call.

SYSTEM STATUS

The UCM6100 system status can be accessed via Web GUI->Status->System Status, which displays the following system information.

General

Network

Storage Usage

Resource Usage

GENERAL

Under Web GUI->Status->System Status->General, users could check the hardware and software information for the UCM6100. Please see details in the following table.

Table 78: System Status->General

Status ->System Status -> General

Model Product model.

Part Number

System Time

Up Time

Product part number.

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.

Boot

Core

Base

Program

Recovery

Boot version.

Core version.

Base version.

Program version. This is the main software release version.

Recovery version.

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NETWORK

Under Web GUI->Status->System Status->Network, users could check the network information for the

UCM6100. Please see details in the following table.

Table 79: 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.

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Figure 103: 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.

Firmware Version 1.0.7.10

Figure 104: System Status->Resource Usage

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SYSTEM EVENTS

The UCM6100 can monitor important system events, log the alerts and send Email notifications to the system administrator.

ALERT EVENTS LIST

The system alert events list can be found under Web GUI->Status->System Events->Alert Events List.

The following event are currently supported on the UCM6100 which will have alert and/or Email generated if occurred:

Disk Usage

Modify Admin Password

Memory Usage

System Reboot

System Update

System Crash

Register SIP Failed

Register SIP Trunk Failed

Restore Config

User Login Success

User Login Failed

SIP Internal Call Failure

SIP Outgoing Call through Trunk Failure

Fail2ban Blocking

SIP Lost Registration

SIP Peer Trunk Status

Click on to configure the parameters for each event. See examples below.

1. Disk Usage.

Firmware Version 1.0.7.10

Figure 105: System Events->Alert Events Lists: Disk Usage

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Detect Cycle: The UCM6100 will perform the internal disk usage detection based on this cycle.

Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.

Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM6100 system will send the alert.

2. Memory Usage

Figure 106: System Events->Alert Events Lists: Memory Usage

Detect Cycle: The UCM6100 will perform the memory usage detection based on this cycle.

Users can enter the number and then select second(s)/minute(s)/hour(s)/day(s) to configure the cycle.

Alert Threshold: If the detected value exceeds the threshold (in percentage), the UCM6100 system will send the alert.

3. System Reboot

Figure 107: System Events->Alert Events Lists: System Reboot

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

4. System Crash

Firmware Version 1.0.7.10

Figure 108: System Events->Alert Events Lists: System Crash

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

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

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

ALERT LOG

Under Web GUI->Status->System Events->Alert Log, system messages are listed when the alert is triggered for the configured system events. The following picture shows disk usage alert log. We can tell the detect cycle for the disk usage is 10 minutes and the disk usage is restored to normal after the administrator cleans up the disk storage below the threshold.

Figure 109: System Events->Alert Log

ALERT CONTACT

Users could add administrator's Email address under Web GUI->Status->System Events->Alert

Contact to send the alert notification to. Up to 10 Email addresses can be added.

CDR

A Call Detail Record (CDR) is a data record produced by telephone exchange activities or other telecommunications equipment documenting the details of a phone call that passed through the PBX. The

CDR is composed of the following data fields on the UCM6100.

Start Time. Format: 2013-03-27 16:47:03.

Call From. Format: "John Doe"<6012>.

Call To. Format: 6005.

Answered By. Format: 6005.

Call Time. Format: 0:00:10.

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Talk Time. Format: 0:00:10

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

Options. Voice record playing/downloading/deleting.

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

Inbound calls

Outbound calls

Internal calls

External calls

Caller Number

Caller Name

From Date

To Date

Figure 110: CDR Filter

Table 80: 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 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.

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The call report will display as the following figure shows.

Figure 111: 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 112: Call Report Entry with Audio Recording File

DOWNLOADED CDR FILE

The downloaded CDR (.csv file) has different format from the web UI CDR. Here are some descriptions.

Call From, Call To

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"Call From": the caller ID.

"Call To": the callee ID.

If "Call From" shows empty, "Call To" shows "s" (see highlight part in the picture below) and the "Source

Channel" contains "DAHDI", this means the call is from FXO/PSTN line. For FXO/PSTN line, we only know there is an incoming request when there is incoming call but we don't know the number being called. So we are using "s" to match it where "s" means "start".

Figure 113: 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 114: Downloaded CDR File Sample - Source Channel and Dest Channel 1

DAHDI means it is an analog call, FXO or FXS.

For UCM6102, DAHDI/(1-2) are FXO ports, and DAHDI(3-4) are FXS ports.

For UCM6104, DAHDI/(1-4) are FXO ports, and DAHDI(5-6) are FXS ports.

For UCM6108, DAHDI/(1-8) are FXO ports, and DAHDI(9-10) are FXS ports.

For UCM6116, DAHDI/(1-16) are FXO ports, and DAHDI/(17-18) are FXS ports.

Sample 2:

Figure 115: Downloaded CDR File Sample - Source Channel and Dest Channel 2

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"SIP" means it's a SIP call. There are three possible format:

(a) SIP/NUM-XXXXXX, where NUM is the local SIP extension number. The last XXXXX is a random string and can be ignored.

(c) SIP/trunk_X/NUM, where trunk_X is the internal trunk name, and NUM is the number to dial out through the trunk.

(c) SIP/trunk_X-XXXXXX, where trunk_X is the internal trunk name and it is an inbound call from this trunk. The last XXXXX is a random string and can be ignored.

Sample 3:

Figure 116: Downloaded CDR File Sample - Source Channel and Dest Channel 3

This is a very special channel name. If it shows up, most likely it means a conference call.

There are some other possible values, but these values are almost the application name which are used by the dialplan.

IAX2/NUM-XXXXXXX: it means this is an IAX call.

Local/@from-internal-XXXXX: it is used internally to do some special feature procedure. We can simply ignore it.

Hangup: the call is hung up from the dialplan. This indicates there are some errors or it has run into abnormal cases.

Playback: play some prompts to you, such as 183 response or run into an IVR.

ReadExten: collect numbers from user. It may occur when you input PIN codes or run into DISA

STATISTICS

CDR Statistics is an additional feature on the UCM6100 which provides users a visual overview of the call report across the time frame. Users can filter with different criteria to generate the statistics chart.

Firmware Version 1.0.7.10

Figure 117: CDR Statistics

UCM6100 Series IP PBX User Manual Page 216 of 239

Trunk Type

Call Type

Time Range

Table 81: 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

This page lists all the recording files recorded by "Auto Record" per extension/ring group/call queue/trunk, or via feature code "Audio Mix Record". If external storage device is plugged in, for example, SD card or

USB drive, the files are stored on the external storage. Otherwise, internal storage will be used on the

UCM6100.

Figure 118: CDR->Recording Files

Click on

“Delete Selected Recording Files” to delete the recording files.

Cl ick on “Delete All Recording Files” to delete all recording files.

Click on to download the recording file in .wav format.

Click on to delete the recording file.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 217 of 239

To sort the recording file, click on the title "Caller", "Callee" or "Call Time" for the corresponding column. Click on the title again can switch the sorting mode between ascending order or descending order.

CDR API CONFIGURATION FILES

The UCM6100 supports third party billing interface API for external billing software to access CDR on the

PBX. The API uses HTTPS to request the CDR data matching given parameters as configured on the third party application. Before accessing the API, the administrators need enable API and configure the access/authentication information on the UCM6100 first.

Table 82: CDR API Configuration Files

Enable Enable/Disable CDR API. The default setting is disabled.

TLS Bind Address Configure the IP address for TLS server to bind to. "0.0.0.0" means binding to all interfaces. The port number is optional and the default port number is 8443. The IP address must match the common name (host name) in the certificate so that the

TLS socket won't bind to multiple IP addresses. The default setting is 0.0.0.0:8443.

TLS Private Key

TLS Cert

Upload TLS private key. The size of the key file must be under 2MB. This file will be renamed as 'private.pem' automatically.

Upload TLS cert. The size of the certificate must be under 2MB. This is the certificate file (*.pem format only) for TLS connection. This file will be renamed as

"certificate.pem" automatically. It contains private key for the client and signed certificate for the server.

TLS Authentication

Name

Configure the user name for TLS authentication. If not configured, authentication will be skipped.

TLS Authentication

Password

Configure the password for TLS authentication. This is optional.

Permitted Specify a list of IP addresses permitted by CDR API. This creates an AIP-specific access control list. Multiple entries are allowed.

For example, "192.168.40.3/255.255.255.255" denies access from all IP addresses except 192.168.40.3.

The format of the HTTPS request for the CDR API is as below.

https://[UCM IP]:[Port]/cdrapi?[option1]=[value]&[option2]=[value]&...

By default, the port number for the API is 8443.

The options included in the request URI control the record matching and output format. For CDR

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matching parameters, all non-empty parameters must have a match to return a record. Parameters can appear in the URI in any order. Multiple values given for caller or callee will be concatenated. The following table shows the parameter list used in the CDR API.

Field format

Value csv, xml, json numRecords Number: 0-1000

Table 83: CDR API URI Parameters

Details

Define the format for output of matching CDR rows.

Default is csv (comma separated values).

Number of records to return. Default is 1000, which is also the maximum allowed value.

offset caller callee startTime endTime

Number

Number of matching records to skip. This will be combined with numRecords to receive all matches over multiple responses. Default is 0.

Comma separated extensions, ranges of extensions, or regular expressions.

Filters based on src (caller) or dst (callee) value, matching any extension contained in the parameter input string.

Example: caller=5300,5302-5304,[email protected]

-OR- caller=5300&caller=5302-

5304&[email protected]

Patterns containing one or more wildcards ('@' or '_') will match as a regular expression, and treat '-' as a literal hyphen rather than a range signifier. The '@' wildcard matches any number of characters (including zero), while '_' matches any single character.

Otherwise, patterns containing a single hyphen will be matching a range of numerical extensions, with nonnumerical characters ignored, while patterns containing multiple hyphens will be ignored. (The pattern "0-0" will match all non-numerical and empty strings).

(Matches extensions 5300,

5302, 5303, 5304, and any extension containing 4 as the second digit/character).

Date and/or time of day in any of the following formats:

YYYY-MM-DDTHH:MM

YYYY-MM-DDTHH:MM:SS

Filters based on the start (call start time) value. Calls which start within this period (inclusive of boundaries) will match, regardless of the call answer or end time. An empty value for either field will be interpreted as range with no minimum or maximum respectively.

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YYYY-MM-DDTHH:MM:SS.SSS

(literal 'T' character separator in above three formats)

HH:MM

Strings without a date have a default value of 2000-01-

01. Strings without a time of day have a default value of of 00:00 UTC, while strings with a time of day specified may also optionally specify a time zone offset - replace

'+' in time zone offset with '%2B' (see http://www.w3.org/TR/NOTE-datetime).

HH:MM:SS

HH:MM:SS.SSS now

DDDDDDDDDD minDur

Number (duration in seconds)

Filters based on the billsec value, the duration between call answer and call end.

maxDur

Example Queries:

The following illustrates the format of queries to accomplish certain requests. In most cases, multiple different queries will accomplish the same goal, and these examples are not intended to be exhaustive, but rather to bring attention to particular features of the CDR API connector.

Query 1: Request all records of calls placed on extension 5300 which last between 8 and 60 seconds

(inclusive), with results in CSV format.

https://192.168.254.200:8088/cdrapi?format=CSV&caller=5300&minDur=8&maxDur=60

-OR-

https://192.168.254.200:8088/cdrapi?caller=5300&minDur=8&maxDur=60

Query 2: Request all records of calls placed on extension 5300 or in the range 6300-6399 to extensions starting with 5, with results in XML format.

https://192.168.254.200:8088/cdrapi?format=XML&caller=5300,6300-6399&[email protected]

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-OR-

https://192.168.254.200:8088/cdrapi?cdrapi?format=XML&caller=5300&caller=6300-6399&[email protected]

Query 3: Request all records of calls placed on extensions containing substring "53" prior to January 23,

2013 00:00:00 UTC to extensions 5300-5309, with results in CSV format.

https://192.168.254.200:8088/[email protected]@&callee=5300-5309&endTime=2013-01-23

-OR-

https://192.168.254.200:8088/[email protected]@&callee=530_&endTime=2013-01-23T00:00:00

Query 4: Request all records of calls placed by an Anonymous caller during July 2013 Central Standard

Time to extensions starting with 2 or 34 or ending with 5, with results in CSV format.

https://192.168.254.200:8088/cdrapi?caller=Anonymous&[email protected],[email protected],@5&startTime=2013-07-01T00:00:00-

06:00&endTime=2013-07-31T23:59:59-06:00

Query 5: Request all records during July 2013 Central Standard Time, 200 at a time, with results in CSV format.

https://192.168.254.200:8088/cdrapi?startTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:59-

06:00&numRecords=200&offset=0

-THEN-

https://192.168.254.200:8088/cdrapi?sstartTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:59-

06:00&numRecords=200&offset=200

-THEN-

https://192.168.254.200:8088/cdrapi?startTime=2013-07-01T00:00:00-06:00&endTime=2013-07-31T23:59:59-

06:00&numRecords=200&offset=400

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Note:

Disallowed characters in the caller, callee, startTime, or endTime strings, and non-digit characters in the values of numRecords, offset, minDur, or maxDur, will result in no records returned - the appropriate container/header for the output format will be the only output. If the format parameter is in error, the CSV header will be used. Error messages will appear in the Asterisk log (along with errors stemming from failed database connections, etc.).

Other errors which return no records include:

- Multiple hyphens in an extension range (e.g. caller=5300-5301-,6300)

- Empty parameter value (e.g. caller=)

- Extension values starting with comma, or with consecutive commas (e.g. caller=5300,,5303)

- Unknown parameters (e.g. caler=5300) or URI ending with '&'

- Except for caller and callee, multiple instances of the same parameter within the URI (e.g. minDur=5&minDur=10)

Example Output:

The following are examples of each of the output formats for the same data set.

CSV:

AcctId,accountcode,src,dst,dcontext,clid,channel,dstchannel,lastapp,lastdata,start,answer,end,duration, billsec,disposition,amaflags,uniqueid,userfield,channel_ext,dstchannel_ext,service

62,,5300,5301,from-internal,"pn01" <5300>,SIP/5300-00000000,SIP/5301-

00000001,Dial,SIP/5301,60,,2013-12-03 11:46:40,2013-12-03 11:46:43,2013-12-03

11:46:49,9,6,ANSWERED,DOCUMENTATION,1386092800.0,EXT,5300,5301,s

63,,5300,5301,from-internal,"pn01" <5300>,SIP/5300-00000000,SIP/5301-

00000001,Dial,SIP/5301,60,,2013-12-03 14:01:41,2013-12-03 14:01:43,2013-12-03

14:01:46,5,3,ANSWERED,DOCUMENTATION,1386100901.0,EXT,5300,5301,s

64,,5300,5301,from-internal,"pn01" <5300>,SIP/5300-00000002,SIP/5301-

00000003,Dial,SIP/5301,60,,2013-12-03 14:02:23,2013-12-03 14:02:27,2013-12-03

14:02:31,8,4,ANSWERED,DOCUMENTATION,1386100943.2,EXT,5300,5301,s

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 222 of 239

XML:

<root>

<cdr><AcctId>62</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext> from-internal</dcontext><clid>&quot;pn01&quot; &lt;5300&gt;</clid><channel>SIP/5300-

00000000</channel><dstchannel>SIP/5301-

00000001</dstchannel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03

11:46:40</start><answer>2013-12-03 11:46:43</answer><end>2013-12-03

11:46:49</end><duration>9</duration><billsec>6</billsec><disposition>ANSWERED</disposition><a maflags>DOCUMENTATION</amaflags><uniqueid>1386092800.0</uniqueid><userfield>EXT</userfie ld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service></ cdr>

<cdr><AcctId>63</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext> from-internal</dcontext><clid>&quot;pn01&quot; &lt;5300&gt;</clid><channel>SIP/5300-

00000000</channel><dstchannel>SIP/5301-

00000001</dstchannel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03

14:01:41</start><answer>2013-12-03 14:01:43</answer><end>2013-12-03

14:01:46</end><duration>5</duration><billsec>3</billsec><disposition>ANSWERED</disposition><a maflags>DOCUMENTATION</amaflags><uniqueid>1386100901.0</uniqueid><userfield>EXT</userfie ld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service></ cdr>

<cdr><AcctId>64</AcctId><accountcode></accountcode><src>5300</src><dst>5301</dst><dcontext> from-internal</dcontext><clid>&quot;pn01&quot; &lt;5300&gt;</clid><channel>SIP/5300-

00000002</channel><dstchannel>SIP/5301-

00000003</dstchannel><lastapp>Dial</lastapp><lastdata>SIP/5301,60,</lastdata><start>2013-12-03

14:02:23</start><answer>2013-12-03 14:02:27</answer><end>2013-12-03

14:02:31</end><duration>8</duration><billsec>4</billsec><disposition>ANSWERED</disposition><a maflags>DOCUMENTATION</amaflags><uniqueid>1386100943.2</uniqueid><userfield>EXT</userfie ld><channel_ext>5300</channel_ext><dstchannel_ext>5301</dstchannel_ext><service>s</service></ cdr>

</root>

JSON:

{

"cdr":

[

{ "AcctId": "62", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal", "clid":

"\"pn01\" <5300>", "channel": "SIP/5300-00000000", "dstchannel": "SIP/5301-00000001", "lastapp":

"Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 11:46:40", "answer": "2013-12-03 11:46:43",

"end": "2013-12-03 11:46:49", "duration": "9", "billsec": "6", "disposition": "ANSWERED", "amaflags":

"DOCUMENTATION", "uniqueid": "1386092800.0", "userfield": "EXT", "channel_ext": "5300",

"dstchannel_ext": "5301", "service": "s" },

{ "AcctId": "63", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal", "clid":

"\"pn01\" <5300>", "channel": "SIP/5300-00000000", "dstchannel": "SIP/5301-00000001", "lastapp":

"Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 14:01:41", "answer": "2013-12-03 14:01:43",

"end": "2013-12-03 14:01:46", "duration": "5", "billsec": "3", "disposition": "ANSWERED", "amaflags":

"DOCUMENTATION", "uniqueid": "1386100901.0", "userfield": "EXT", "channel_ext": "5300",

"dstchannel_ext": "5301", "service": "s" },

{ "AcctId": "64", "accountcode": "", "src": "5300", "dst": "5301", "dcontext": "from-internal", "clid":

"\"pn01\" <5300>", "channel": "SIP/5300-00000002", "dstchannel": "SIP/5301-00000003", "lastapp":

"Dial", "lastdata": "SIP/5301,60,", "start": "2013-12-03 14:02:23", "answer": "2013-12-03 14:02:27",

"end": "2013-12-03 14:02:31", "duration": "8", "billsec": "4", "disposition": "ANSWERED", "amaflags":

}

"DOCUMENTATION", "uniqueid": "1386100943.2", "userfield": "EXT", "channel_ext": "5300",

"dstchannel_ext": "5301", "service": "s" }

]

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 223 of 239

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

Firmware Server Path

Firmware Version 1.0.7.10

Figure 119: Network Upgrade

Table 84: Network Upgrade Configuration

Allow users to choose the firmware upgrade method: TFTP, HTTP or

HTTPS.

Define the server path for the firmware server.

UCM6100 Series IP PBX User Manual Page 224 of 239

Firmware File Prefix If configured, only the firmware with the matching encrypted prefix will be downloaded and flashed into the UCM6100.

Firmware File Suffix

HTTP/HTTPS User Name

HTTP/HTTPS Password

If configured, only the firmware with the matching encrypted postfix will be downloaded and flashed into the UCM6100.

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 120: Local Upgrade

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 225 of 239

Figure 121: Upgrading Firmware Files

Wait until the upgrading process is successful and a window will be popped up in the Web GUI.

Figure 122: Reboot UCM6100

Click on "OK" to reboot the UCM6100 and check the firmware version after it boots up.

Note:

Please do not interrupt or power cycle the UCM6100 during upgrading process.

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NO LOCAL FIRMWARE SERVERS

For users that would like to use remote upgrading without a local TFTP server, Grandstream offers a NATfriendly HTTP server. This enables users to download the latest software upgrades for their devices via this server. Please refer to the webpage: http://www.grandstream.com/support/firmware .

Alternatively, users can download a free TFTP or HTTP server and conduct a local firmware upgrade. A free windows version TFTP server is available for download from : http://www.solarwinds.com/products/freetools/free_tftp_server.aspx

http://tftpd32.jounin.net

Instructions for local firmware upgrade via TFTP:

1. Unzip the firmware files and put all of them in the root directory of the TFTP server;

2. Connect the PC running the TFTP server and the UCM6100 to the same LAN segment;

3. Launch the TFTP server and go to the File menu->Configure->Security to change the TFTP server's default setting from "Receive Only" to "Transmit Only" for the firmware upgrade;

4. Start the TFTP server and configure the TFTP server in the UCM6100 web configuration interface;

5. Configure the Firmware Server Path to the IP address of the PC;

6. Update the changes and reboot the UCM6100.

End users can also choose to download a free HTTP server from http://httpd.apache.org/ or use

Microsoft IIS web server.

BACKUP

The UCM6100 configuration can be backed up locally or via network. The backup file will be used to restore the configuration on UCM6100 when necessary.

LOCAL BACKUP

Users could backup the UCM6100 configurations for restore purpose under Web GUI->Maintenance-

>Backup->Local Backup. Before creating new backup file, select the backup option first.

If the Config-File is selected only, the backup file will be saved in the flash of the UCM6100.

If Voice-File, Voicemail-File, Voice-Records, CDR or VFAX is selected, external storage devices (USB

Flash drive or SD Card) will be required because the backup file might be too large.

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Click on "Create New Backup" button to start backup. Once the backup is done, the list of the backups will be displayed with date and time in the web page. Users can download , restore , or delete it from the UCM6100 internal storage or the external device.

Figure 123: 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.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 228 of 239

Figure 124: Data Sync

Table 85: 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 restore from the backup file.

to

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 229 of 239

Figure 125: 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.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 230 of 239

Enable CDR Cleaner

CDR Clean Time

Clean Interval

Enable VR Cleaner

Figure 126: Cleaner

Table 86: 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.

Choose Cleaner File

VR Clean Threshold

VR Clean Time

Clean Interval

Select the files for system automatic clean.

Recording Files

Conference

Queue

Voicemail

Fax

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.

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.

Firmware Version 1.0.7.10 UCM6100 Series IP PBX User Manual Page 231 of 239

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 127: Reset and Reboot

SYSLOG

On the UCM6100, users could dump the syslog information to a remote server under Web GUI-

>Maintenance->Syslog. Enter the syslog server hostname or IP address and select the module/level for the syslog information.

The default syslog level for all modules is "error", which is recommended in your UCM6100 settings because it can be helpful to locate the issues when errors happen.

Some typical modules for UCM6100 functions are as follows and users can turn on "notic" and "verb" levels besides "error" level. pbx: This module is related to general PBX functions. chan_sip: This module is related to SIP calls. chan_dahdi: This module is related to analog calls (FXO/FXS). app_meetme: This module is related to conference bridge.

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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 128: Ethernet Capture

The output result is in .pcap format. Therefore, users could specify the capture filter as used in general network traffic capture tool (host, src, dst, net, protocol, port, port range) before starting capturing the trace.

IP PING

Enter the target host in host name or IP address. Then press "Start" button. The output result will dynamically display in the window below.

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Figure 129: 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 130: Traceroute

ANALOG RECORD TRACE

Analog record trace can be used to troubleshoot analog trunk issue, for example, the UCM6100 user has

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caller ID issue for incoming call from Analog trunk. Users can access analog record trance under web

GUI->Maintenance->Troubleshooting ->Analog Record Trace.

Here is the step to capture trace:

1. Select FXO or FXS for "Record Ports". If the issue happens on FXO 1, select FXO port 1 to record the trace.

2. Select "Record Direction".

3. Select "Record File Mode" to separate the record per direction or mix.

4. Click on "Start".

5. Make a call via the analog port that has the issue.

6. Once done, click on "Stop".

7. Click on "Download" to download the analog record trace.

Figure 131: Troubleshooting Analog Trunks

After capturing the trace, users can download it for basic analysis. Or you can contact Grandstream

Technical support in the following link for further assistance if the issue is not resolved. http://www.grandstream.com/index.php/support

SERVICE CHECK

Enable Service Check to periodically check Asterisk service. Check Cycle is configurable in seconds and the default setting is 60 sec. Check Times is the maximum number of failed checks before restart the Asterisk service. The default setting is 3. If there is no response from Asterisk service after 3 attempts (default) to check, current status will be stored and Asterisk service will be restarted.

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Figure 132: Service Check

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EXPERIENCING THE UCM6100 SERIES IP PBX

Please visit our website: http://www.grandstream.com

to receive the most up- to-date updates on firmware releases, additional features, FAQs, documentation and news on new products.

We encourage you to browse our product related documentation , FAQs and User and Developer Forum for answers to your general questions. If you have purchased our products through a Grandstream

Certified Partner or Reseller, please contact them directly for immediate support.

Our technical support staff is trained and ready to answer all of your questions. Contact a technical support member or submit a trouble ticket online to receive in-depth support.

Thank you again for purchasing Grandstream UCM6100 series IP PBX appliance, it will be sure to bring convenience and color to both your business and personal life.

* Asterisk is a Registered Trademark of Digium, Inc.

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FCC Caution:

Any Changes or modifications not expressly approved by the party responsible for compliance could void the user's authority to operate the equipment.

This device complies with part 15 of the FCC Rules. Operation is subject to the following two conditions:

(1) This device may not cause harmful interference, and (2) this device must accept any interference received, including interference that may cause undesired operation.

Note: This equipment has been tested and found to comply with the limits for a Class B digital device, pursuant to part 15 of the FCC Rules. These limits are designed to provide reasonable protection against harmful interference in a residential installation. This equipment generates, uses and can radiate radio frequency energy and, if not installed and used in accordance with the instructions, may cause harmful interference to radio communications. However, there is no guarantee that interference will not occur in a particular installation. If this equipment does cause harmful interference to radio or television reception, which can be determined by turning the equipment off and on, the user is encouraged to try to correct the interference by one or more of the following measures:

- Reorient or relocate the receiving antenna.

- Increase the separation between the equipment and receiver.

- Connect the equipment into an outlet on a circuit different from that to which the receiver is connected.

- Consult the dealer or an experienced radio/TV technician for help.

Regulatory Information

U.S. FCC Part 68 Statement

This equipment complies with Part 68 of the FCC rules and the requirements adopted by the ACTA. The unit bears a label on the back which contains among other information a product identifier in the format

US: GNIIS00BUCM6104/US: GNIIS00BUCM6116. If requested, this number must be provided to the telephone company.

This equipment uses the following standard jack types for network connection: RJ11C.

This equipment contains an FCC compliant modular jack. It is designed to be connected to the telephone network or premises wiring using compatible modular plugs and cabling which comply with the requirements of FCC Part 68 rules.

The Ringer Equivalence Number, or REN, is used to determine the number of devices which may be connected to the telephone line. An excessive REN may cause the equipment to not ring in response to an incoming call. In most areas, the sum of the RENs of all equipment on a line should not exceed five

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(5.0).

In the unlikely event that this equipment causes harm to the telephone network, the telephone company can temporarily disconnect your service. The telephone company will try to warn you in advance of any such disconnection, but if advance notice isn't practical, it may disconnect the service first and notify you as soon as possible afterwards. In the event such a disconnection is deemed necessary, you will be advised of your right to file a complaint with the FCC.

From time to time, the telephone company may make changes in its facilities, equipment, or operations which could affect the operation of this equipment. If this occurs, the telephone company is required to provide you with advance notice so you can make the modifications necessary to obtain uninterrupted service.

There are no user serviceable components within this equipment. See Warranty flyer for repair or warranty information.

It shall be unlawful for any person within the United States to use a computer or other electronic device to send any message via a telephone facsimile unless such message clearly contains, in a margin at the top or bottom of each transmitted page or on the first page of the transmission, the date and time it is sent and an identification of the business, other entity, or individual sending the message and the telephone number of the sending machine or of such business, other entity, or individual. The telephone number provided may not be a 900 number or any other number for which charges exceed local or long distance transmission charges. Telephone facsimile machines manufactured on and after December 20, 1992, must clearly mark such identifying information on each transmitted message. Facsimile modem boards manufactured on and after December 13, 1995, must comply with the requirements of this section.

This equipment cannot be used on public coin phone service provided by the telephone company.

Connection to Party Line Service is subject to state tariffs. Contact your state public utility commission, public service commission, or corporation commission for more information.

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